Method and apparatus for extracting a desired television signal from a wideband IF input using re-sampling

REFERENCE TO RELATED APPLICATIONS

0001This application is a continuation of the following application, U.S. patent application Ser. No. 12/041,685, entitled METHOD AND APPARATUS FOR EXTRACTING A DESIRED TELEVISION SIGNAL FROM A WIDEBAND IF INPUT, filed on Mar. 4, 2008, which is a continuation-in-part of Ser. No. 12/038,781 filed Feb. 27, 2008 which both claim priority from U.S. Provisional Patent Application Ser. No. 60/894,832 filed on Mar. 14, 2007, and all three of which are hereby incorporated by reference as if set forth in full in this application for all purposes.

FIELD

0002Various embodiments of systems, system blocks and corresponding methods are described herein that relate to a universal television receiver that can process television signals that are broadcast according to different television broadcast standards.

BACKGROUND

0003Television signals are broadcast according to several different types of television broadcast standards. These television broadcast standards include variations of NTSC, SECAM and PAL for analog signals, and ATSC, DVB-T and ISDB-T for digital signals. These television broadcast standards have different characteristics such as bandwidth, modulation type and the location of audio in the case of analog signals. Accordingly, traditional television receivers have been specifically built to process certain television signals based on a particular television broadcast standard. Traditional television receivers typically use a SAW filter that has a very sharp frequency response with a passband that corresponds to the bandwidth of the television signal that is being received. The SAW filter is a relatively expensive component that cannot be integrated onto a chip, and does not readily allow a television receiver to receive television signals that are transmitted according to different television broadcast standards, without further specialized processing.

SUMMARY

0004In one aspect of the invention, at least one of the embodiments described herein provides a television receiver for processing television signals to provide video and audio information for a desired television channel signal, the television signals being transmitted according to a variety of television broadcast standards. The television receiver comprises an analog processing block for filtering and amplifying a multi-channel television signal to produce a coarse channel signal; an analog to digital converter (ADC) coupled to the analog processing block for digitizing the coarse channel signal to produce a digitized coarse channel signal; and a digital processing block coupled to the ADC for processing the digitized coarse channel signal to obtain video and audio information for the desired television channel signal. For analog and digital television broadcast standards, the processing comprises re-sampling a processed version of the digitized coarse channel signal at a first new sampling rate, with respect to the sampling rate employed by the ADC, to adjust a normalized bandwidth of the desired television channel signal to generally correspond to the normalized passband of a video filter.

0005In another aspect of the invention, at least one of the embodiments described herein provides a method for processing television signals to provide video and audio information for a desired television channel signal, the television signals being transmitted according to a variety of television broadcast standards. The method comprises filtering and amplifying a multi-channel television signal to produce a coarse channel signal; digitizing the coarse channel signal to produce a digitized coarse channel signal; and processing the digitized coarse channel signal to obtain video and audio information for the desired television channel signal. For analog and digital television broadcast standards, the processing comprises re-sampling a processed version of the digitized coarse channel signal at a first new sampling rate, with respect to the sampling rate employed during digitization, to adjust a normalized bandwidth of the desired television channel signal to generally correspond to the normalized passband of a video filter.

BRIEF DESCRIPTION OF THE DRAWINGS

0006For a better understanding of the various embodiments described herein and to show more clearly how they may be carried into effect, reference will now be made, by way of example only, to the accompanying drawings in which:

0007<figref idref="DRAWINGS">FIG. 1A</figref> is a spectral diagram of the entire television band;

0008<figref idref="DRAWINGS">FIG. 1B</figref> is a diagram of a blocking profile showing an exemplary range of signal strength for received television signals;

0009<figref idref="DRAWINGS">FIG. 2</figref> is a high-level block diagram of an exemplary embodiment of a universal television receiver;

0010<figref idref="DRAWINGS">FIG. 3</figref> is a block diagram of an exemplary embodiment of an RF processing block that can be used in the universal television receiver;

0011<figref idref="DRAWINGS">FIG. 4</figref> is a block diagram of an exemplary embodiment of an analog processing block that can be used in the universal television receiver;

0012<figref idref="DRAWINGS">FIGS. 5A-5C</figref> are spectral plots of signals at various locations in the analog processing block of <figref idref="DRAWINGS">FIG. 4</figref>;

0013<figref idref="DRAWINGS">FIGS. 6A-6B</figref> are spectral plots of exemplary filter transfer functions that can be used for filters in the analog processing block;

0014<figref idref="DRAWINGS">FIG. 7</figref> is a block diagram of an exemplary embodiment of a digital processing block that can be used in the universal television receiver;

0015<figref idref="DRAWINGS">FIG. 8</figref> is a flow chart diagram of an exemplary embodiment of a standards detection method for detecting television transmission broadcast standards;

0016<figref idref="DRAWINGS">FIG. 9</figref> is a block diagram of an exemplary embodiment of an input filtering block that can be used in the digital processing block;

0017<figref idref="DRAWINGS">FIG. 10</figref> is a block diagram of an exemplary embodiment of a video processing block that can be used in the digital processing block;

0018<figref idref="DRAWINGS">FIGS. 11A</figref>, <b>11</b>B and <b>11</b>C are spectral diagrams of a general desired television channel signal, a desired television channel signal according to an analog television broadcast standard and a desired television channel signal according to a digital television broadcast standard respectively;

0019<figref idref="DRAWINGS">FIGS. 11D</figref>, <b>11</b>E and <b>11</b>F show spectral plots illustrating the operation of a video pre-polyphase filter, a video polyphase filter and a main video filter that are used in the video processing block;

0020<figref idref="DRAWINGS">FIGS. 11G</figref>, <b>11</b>H and <b>11</b>I show the magnitude, real and imaginary parts of a signal that is processed by a VSB filter of the video processing block;

0021<figref idref="DRAWINGS">FIGS. 11J</figref>, <b>11</b>K and <b>11</b>L show the magnitude, real and imaginary parts of the frequency response of the VSB filter of the video processing block;

0022<figref idref="DRAWINGS">FIG. 12A</figref> is a block diagram of an exemplary embodiment of a carrier recovery block that can be used in the video processing block;

0023<figref idref="DRAWINGS">FIG. 12B</figref> is a diagram illustrating the phenomenon of overmodulation;

0024<figref idref="DRAWINGS">FIG. 12C</figref> is a diagram illustrating a first technique for dealing with overmodulation;

0025<figref idref="DRAWINGS">FIG. 12D</figref> is a diagram illustrating a second technique for dealing with overmodulation;

0026<figref idref="DRAWINGS">FIG. 13A</figref> is a block diagram of an exemplary embodiment of an audio filtering block that can be used in the digital processing block;

0027<figref idref="DRAWINGS">FIG. 13B</figref> is a block diagram of an alternative exemplary embodiment of an audio filtering block that can be used in the digital processing block;

0028<figref idref="DRAWINGS">FIG. 14</figref> is a block diagram of an exemplary embodiment of an audio processing block that can be used in the digital processing block;

0029<figref idref="DRAWINGS">FIG. 15</figref> is a flow chart diagram of an exemplary embodiment of a gain control method that may be employed by the universal television receiver of <figref idref="DRAWINGS">FIG. 3</figref>;

0030<figref idref="DRAWINGS">FIG. 16A</figref> is a block diagram of an exemplary embodiment of the analog gain control block of <figref idref="DRAWINGS">FIG. 9</figref>, which can be used to employ an alternate gain control method;

0031<figref idref="DRAWINGS">FIG. 16B</figref> is a block diagram of a leaky peak detector shown in <figref idref="DRAWINGS">FIG. 16A</figref>;

0032<figref idref="DRAWINGS">FIG. 16C</figref> is an illustration showing RF/IF take-over for the gain control method used by the analog gain control block of <figref idref="DRAWINGS">FIG. 16A</figref>;

0033<figref idref="DRAWINGS">FIG. 17A</figref> is a block diagram of another exemplary embodiment of a universal television receiver;

0034<figref idref="DRAWINGS">FIG. 17B</figref> is a block diagram of an exemplary embodiment of a digital television demodulator;

0035<figref idref="DRAWINGS">FIG. 18</figref> is a block diagram of another exemplary embodiment of a universal television receiver;

0036<figref idref="DRAWINGS">FIG. 19A</figref> is a block diagram of another exemplary embodiment of a universal television receiver that employs aliasing avoidance;

0037<figref idref="DRAWINGS">FIG. 19B</figref> is a block diagram of an exemplary embodiment for the variable Phase Lock Loop of <figref idref="DRAWINGS">FIG. 19A</figref>;

0038<figref idref="DRAWINGS">FIG. 19C</figref> is a spectral plot illustrating interference of a desired television channel due to aliasing;

0039<figref idref="DRAWINGS">FIG. 19D</figref> is a spectral plot illustrating the avoidance of an aliased interferer onto the desired television channel by using sampling rate adjustment;

0040<figref idref="DRAWINGS">FIG. 20A</figref> is a spectral plot illustrating interference of a desired television channel due to mixing distortion products;

0041<figref idref="DRAWINGS">FIG. 20B</figref> is a spectral plot illustrating the avoidance of distortion interference of the desired television channel by using a local oscillator frequency shift;

0042<figref idref="DRAWINGS">FIG. 21A</figref> is another spectral plot illustrating interference of a desired television channel due to mixing of distortion products; and

0043<figref idref="DRAWINGS">FIG. 21B</figref> is a spectral plot illustrating the avoidance of distortion interference of the desired television channel by using a local oscillator frequency shift and sampling rate adjustment.

DETAILED DESCRIPTION

0044It will be appreciated that for simplicity and clarity of illustration, elements shown in the figures have not necessarily been drawn to scale. Further, where considered appropriate, reference numerals may be repeated among the figures to indicate corresponding or analogous elements. In addition, numerous specific details are set forth in order to provide an adequate understanding for practicing the various embodiments described herein. However, it will be understood by those of ordinary skill in the art that the various embodiments described herein may be practiced without these specific details. In other instances, some methods, procedures and components have not been described in detail since they are well known to those skilled in the art. Furthermore, it should be understood that the word “exemplary” is used herein to denote an example embodiment of a device or method and not necessarily indicate a preferred implementation of a device or method.

0045Referring now to <figref idref="DRAWINGS">FIG. 1A</figref>, shown therein are the spectral characteristics of a wideband television signal <b>10</b>. The wideband television signal <b>10</b> is essentially continuous with a positive component ranging from 42 to 862 MHz and a corresponding negative component ranging from −42 to −862 MHz. An individual television signal has a bandwidth in the range of 6-8 MHz depending on the television broadcast standard with which the television signal corresponds. For instance, NTSC television signals that are used in North America have a bandwidth of 6 MHz, while television signals that are used elsewhere may have a bandwidth of 7 or 8 MHz. In some countries, different bandwidths can be used in different parts of the television band. Other parameters will also vary for different television broadcast standards as is commonly known to those skilled in the art.

0046Television signal quality can be dictated by differences in the power of the television signals that are received at the television receiver. The difference in power depends on the local geography and the location of the transmitters with respect to the television receiver. A measure of good television signal quality can be specified in terms of SNR which depends on the television broadcast standard. For instance, analog NTSC television signals may benefit from greater than 45 dB SNR for good signal quality for a Composite Video Baseband Signal (CVBS) output. Meanwhile, digital television signals may require as little as 15 dB carrier-to-noise ratio for good signal quality in the case of ATSC, since processing of digital television signals uses error detection and correction.

0047Referring now to <figref idref="DRAWINGS">FIG. 1B</figref>, shown therein is a television signal blocking profile <b>20</b>, based on the US ATSC A/74 Receiver Performance Guidelines, which is expected to be a widely accepted standard. <figref idref="DRAWINGS">FIG. 1B</figref> shows that there can be a wide range in terms of the strength of a desired television channel signal <b>22</b>, relative to the strength of other channels at nearby frequencies. For instance, according to the ATSC performance guidelines for digital receivers, the weakest signal strength for the desired television channel signal <b>22</b> that should be correctly received is −83 dBm, while the strongest signal strength is −4 dBm. Furthermore, the desired television channel signal <b>22</b> may have strong undesired television channel signals <b>24</b> and <b>26</b> directly adjacent with a relative power of 33 dB. Subsequent adjacent television channels <b>28</b> to <b>46</b> can increase in power by 4 dB for each additional channel separation until 57 dB is reached, at which point reception should still be possible. Conventional television receivers deal with this technical challenge in part by filtering a major portion of the received television signal by using a tracking filter right at the input of the tuning stage of the television receiver.

0048Referring now to <figref idref="DRAWINGS">FIG. 2</figref>, shown therein is a high-level block diagram of an exemplary embodiment of a universal television receiver <b>100</b> that can receive and process digital and analog television signals that are transmitted according to a variety of broadcast signals, and therefore have different television channel signal bandwidths. The universal television receiver <b>100</b> comprises an RF processing block <b>102</b>, an analog processing block <b>104</b>, an analog to digital converter (ADC) <b>106</b>, a digital processing block <b>108</b>, and a digital to analog converter (DAC) block <b>110</b>. The universal television receiver <b>100</b> receives the wideband television signal <b>10</b> and provides a processed version of the desired television channel signal <b>112</b>. The digital processing block <b>108</b> provides control signals to the RF and analog processing blocks <b>102</b> and <b>104</b> and the ADC <b>106</b> as is described in further detail below. Furthermore, depending on the television broadcast standard for the desired television channel signal, the digital processing block <b>108</b> can output a modulated digital signal <b>112</b>′. The DAC block <b>110</b> can contain several digital to analog converters depending on the type of output that is required. For instance, there can be one DAC for a CVBS output and there can be at least one more DAC for a sound IF output. Alternatively, in other embodiments, some output signals provided by the universal television receiver <b>100</b> may be provided to a direct digital connection on a downstream component in which case there is no need for a DAC for these output signals.

0049The RF processing block <b>102</b>, analog processing block <b>104</b> and digital processing block <b>108</b> are custom blocks. However, in an alternative embodiment, an off-the-shelf RF processing block can be used with corresponding changes in some operating parameters and processing in the analog and digital processing blocks <b>104</b> and <b>108</b> and the ADC <b>106</b>. This alternative embodiment is described in further detail below with respect to <figref idref="DRAWINGS">FIGS. 17A and 17B</figref>. The ADC <b>106</b> can have 12 bits of SNR and −72 dBc of linearity for full-scale signals.

0050The universal television receiver <b>100</b> does not include a SAW filter. Rather, the universal television receiver <b>100</b> uses distributed filtering, in both the analog and digital domains, to isolate a desired television channel signal <b>22</b>. This approach allows for the complete realization of the universal television receiver <b>100</b> on a single Integrated Circuit (IC) as well as being able to implement the television receiver <b>100</b> using reduced performance constraints for various processing components when a SAW filter is not used. Accordingly, filters can be implemented with reduced order and reduced Q, thereby requiring less area when realized on an IC. Furthermore, such a design is less sensitive to process variations and operating conditions that vary, such as temperature and voltage. However, in some alternative embodiments a SAW filter can also be used.

0051Referring now to <figref idref="DRAWINGS">FIG. 3</figref>, shown therein is a block diagram of an exemplary embodiment of the RF processing block <b>102</b>. The RF processing block <b>102</b> includes an antenna <b>120</b>, a low noise amplifier (LNA) <b>122</b>, a first variable gain amplifier (VGA) <b>124</b>, a power meter <b>126</b>, a mixing stage <b>128</b> having a mixer <b>130</b> and a frequency synthesizer <b>132</b>, and a second variable gain amplifier (VGA) <b>134</b>. The power meter <b>126</b> is optional depending on the automatic gain control method that is employed which is described in further detail below. The RF processing block <b>102</b> receives and processes the wideband television signal <b>10</b> to provide a multi-channel television signal <b>136</b> that can include on the order of tens of television channel signals including the desired television channel signal <b>22</b>. Accordingly, the RF processing block <b>102</b> provides a first level of filtering, as well as gain or attenuation, as the case may be. In alternative embodiments, a cable connection, a satellite dish or other wireless connection can be used instead of the antenna <b>120</b>.

0052The wideband television signal <b>10</b> is received by the antenna <b>120</b> and then amplified by the LNA <b>122</b> and the VGA <b>124</b>. The amount of gain that is provided by the LNA <b>122</b> and the VGA <b>124</b> is variable based on certain properties of the wideband television signal <b>10</b>, and the desired television channel signal <b>22</b>, which can vary widely. However, in some cases the received wideband television signal <b>10</b> may have to be attenuated. In general, the combined amount of gain provided by the LNA <b>122</b> and the VGA <b>124</b> can vary from −20 dB to 50 dB. Accordingly, both of the amplifiers <b>122</b> and <b>124</b> have a wide dynamic amplitude range. The amount of gain provided by the VGA <b>124</b> is controlled by AGC control signal <b>138</b>. Although signal <b>138</b> is shown as analog, it can be either analog or digital.

0053The frequency response of the LNA <b>122</b> and the VGA <b>124</b> can extend up to about 1 GHz to pass the received television signal along with some harmonics without significant distortion. However, in alternative embodiments, at least one additional filter (i.e. a switchable filter, a tracking filter, an FM notch filter, or an FM band filter for improved FM performance) can be inserted before the LNA <b>122</b> to remove unwanted signals such as cell phone and short-wave radio signals. Furthermore, FM radio signals can also be filtered out. Alternatively, FM radio signals can be included if radio functionality is also desired. In alternative embodiments, the LNA <b>122</b> and the VGA <b>124</b> can be combined and implemented in one variable gain stage.

0054The mixing stage <b>128</b> mixes the output of the VGA <b>124</b> to a much higher frequency range on the order of GHz. At this higher frequency range, it is easier to implement the components of the analog processing block <b>104</b>. In addition, harmonics of the mixing stage <b>128</b> are at very high frequencies thus avoiding any potential overlap with the television band that could result in any interfering images. Also, in this frequency range interference from other television signals can be minimized due to various signal processing techniques used in the analog processing block <b>104</b>. For example, the mixing stage <b>128</b> can provide a high-side mix to mix the output of the VGA <b>124</b> so that the desired television channel signal <b>22</b> is centered near 1.125 GHz (see <figref idref="DRAWINGS">FIG. 5A</figref>). To achieve this, the frequency synthesizer <b>132</b> is tunable and, in some cases, can provide a tunable signal with a frequency in the range of 1.16 GHz to 1.9 GHz. For example, with an oscillation frequency of 1.167 GHz, the low end of the television band (i.e. −42 MHz) appears at 1.125 GHz, and with an oscillation frequency of 1.987 GHz, the high end of the television band (i.e. −862 MHz) appears at 1.125 GHz. The frequency synthesizer <b>132</b> receives a tuning control signal <b>140</b> from the digital processing block <b>108</b> based on the frequency of the desired television channel signal <b>22</b> and the frequency at which the frequency content of the desired television channel signal <b>22</b> is to be placed for processing by the analog processing block <b>104</b>.

0055The frequency synthesizer <b>132</b> acts as a Local Oscillator (LO) and can be based on a PLL design and the oscillation signal can be derived from a crystal oscillator having a frequency of 4 MHz or 16 MHz for example. The frequency synthesizer <b>132</b> will have a certain amount of offset error that will need to be resolved by the digital processing block <b>108</b> to locate the desired television channel signal <b>22</b>. However, due to coarse filtering that is provided in the analog processing block <b>104</b> and the signal processing provided by the digital processing block <b>108</b>, the frequency synthesizer <b>132</b> can be realized with a coarser or larger step size such that there is a larger amount of offset in the exact location of the desired television channel signal <b>22</b> after mixing (i.e. a larger shift from being centered at 1.125 GHz). Generally, the universal television receiver <b>100</b> can tolerate offsets of 1 MHz or more, particularly when the offset is due to the use of a coarse synthesizer step size, since the offset in this case is known and can be compensated further downstream in the processing chain. This is discussed in further detail below.

0056The mixing stage <b>128</b> can also provide some gain, and filtering. In some implementations, the mixer <b>130</b> can include a differential gain stage that can provide about 10-20 dB of gain, and can include a reactive load, such as a pair of LC tank filters, to provide filtering around 1.125 GHz. The filtering is not sharp since the inductors are realized on an IC, and the Q of the LC tank filters can range from 6 to 12 at certain frequencies. Accordingly, the bandwidth of the filtering provided in the mixing stage <b>128</b> can be on the order of one hundred MHz, and the output of the mixing stage <b>128</b> can include over ten television channels. Alternatively, bond-wires can be used for the inductors to achieve Q-values of up to 10 or greater. In at least some cases, external inductors can also be used for greater selectivity.

0057The output of the mixing stage <b>128</b> is then amplified by the VGA <b>134</b>, which can be a standard VGA. The VGA <b>134</b> can generally be used to provide about 10 to 30 dB of gain. The amount of gain provided by the VGA <b>134</b> is selected based on the gain provided by the VGA <b>124</b>, as well as the amount of gain and filtering that is provided by the mixing stage <b>128</b>. Gain control for the VGAs <b>124</b> and <b>134</b> is described in further detail below. In alternative embodiments, if there is no filtering after the VGA <b>124</b> or if no additional gain is necessary, then the VGA <b>134</b> is optional and can be excluded.

0058Referring now to <figref idref="DRAWINGS">FIG. 4</figref>, shown therein is a block diagram for an exemplary embodiment of the analog processing block <b>104</b>. The analog processing block <b>104</b> generally filters and amplifies a multi-channel television signal <b>136</b> to produce a coarse channel signal <b>162</b>. The filtering is referred to as coarse filtering in that bandwidths are used for the filters that are large enough to accommodate different bandwidths for the desired television channel signal <b>22</b> due to the various different types of analog and digital broadcast standards that are used. The bandwidths are also selected to be wide or large enough to accommodate any frequency shifts or offsets in the desired television channel signal <b>22</b> as well as any variability due to analog circuitry. This variability includes component tolerances, temperature and voltage variations (which can result in a variation in absolute frequency and the bandwidth of the signal path), irregularities which occur near the band edges of the filters (using a wide band minimizes the effects when not pushing the edges) and easing the difficulty of precise analog design (more specifically, the tuning required to set and keep the band pass of the filters precise). These variations in frequency can also be tolerated by tracking the carrier frequency of the desired television channel signal <b>22</b> for both analog and digital broadcast standards. For analog broadcast standards, the carrier tracking is akin to tracking the picture carrier and audio carrier for the desired television channel (although in some embodiments audio carrier tracking can be slaved to picture carrier tracking as discussed below with regards to <figref idref="DRAWINGS">FIG. 13B</figref>). For digital broadcast standards, the carrier tracking is optional but when performed is akin to tracking a specified frequency such as a center frequency. For instance, even though DVB-T signals can be considered to have as many as 8192 carriers, tuning can be done by specifying a center frequency for the 8192 carrier frequencies and tracking the middle carrier frequency. The carrier frequency tracking is discussed further with regards to <figref idref="DRAWINGS">FIGS. 10</figref>, <b>12</b>A, <b>13</b>A, <b>13</b>B, <b>17</b>A and <b>17</b>B. When performed, the carrier tracking is used to introduce a frequency shift feedback signal to ensure that the desired television channel signal <b>22</b> remains in the bandwidth of the filtering components in the digital processing block <b>108</b>. This is described in more detail with regards to <figref idref="DRAWINGS">FIG. 10</figref>.

0059The analog processing block <b>104</b> includes a first coarse bandpass filter <b>150</b>, a third VGA <b>152</b>, a sample and hold circuit <b>154</b>, a discrete-time coarse bandpass filter <b>156</b> (which can be based on a switched-capacitor implementation), a discrete-time VGA <b>158</b> and a frequency synthesizer <b>160</b>. Gain control for the VGAs <b>152</b> and <b>158</b> is discussed in more detail further below. In some cases, one of the VGAs <b>152</b> and <b>158</b> is optional and can be excluded in alternative embodiments. If the coarse bandpass filter <b>150</b> is not present, then the VGA <b>152</b> is not required. If the discrete-time bandpass filter <b>156</b> is not required, then the VGA <b>158</b> will not be necessary. In addition, the coarse bandpass filter <b>150</b> can be implemented in a discrete or integrated fashion. In an alternative embodiment of the analog processing block <b>104</b>, if a continuous-time bandpass sigma-delta ADC is used, then the sample-and-hold circuit <b>154</b> is not required and the ADC has inherent anti-alias filtering potentially precluding the need for other filters. Also the coarse bandpass filter <b>156</b> is a continuous-time filter and the VGA <b>158</b> is a continuous time variable gain amplifier.

0060Generally, the analog processing block <b>104</b> processes the multi-channel television signal <b>136</b> to provide a coarse channel signal <b>162</b> that includes the frequency content of the desired television channel signal <b>22</b> as well as portions, or the entirety, of one or more adjacent television channel signals. In some implementations, the coarse channel signal <b>162</b> can have a bandwidth in the range of 10-20 MHz and in some cases can be approximately 10 MHz, and therefore, for some television broadcast standards can include one full channel and two partial channels (see <figref idref="DRAWINGS">FIG. 5B</figref> for example) or two full television channel signals. The analog processing block <b>104</b> utilizes distributed, coarse filtering to provide the coarse channel signal <b>162</b> with enough bandwidth to address the various issues mentioned previously. This is described in further detail below.

0061The coarse bandpass filter <b>150</b> provides another level of filtering to remove unwanted television channel signals as well as to prevent aliasing due to subsequent discrete time sampling. The coarse bandpass filter <b>150</b> is also approximately centered at the frequency to which the mixing stage <b>128</b> mixes the desired television channel signal <b>22</b>. The size of the passband of the coarse bandpass filter <b>150</b> is large enough to pass a coarse frequency region of interest <b>170</b><i>c </i>(see <figref idref="DRAWINGS">FIG. 5C</figref>), which includes the desired television channel signal <b>22</b> and at least portions of at least one or more adjacent television channel signals.

0062The sampling rate of the sample and hold circuit <b>154</b> and the bandwidth of the coarse bandpass filter <b>150</b> can be selected to employ sub-sampling so that another mixer is not needed to shift the coarse frequency region of interest <b>170</b><i>c </i>to IF. The sampling rate is selected to be more than twice the bandwidth of the coarse frequency region of interest <b>170</b><i>c</i>, which can generally be about 10-20 MHz wide. However, it is not practical for the coarse bandpass filter <b>150</b> to have a bandwidth of 10-20 MHz with a center frequency in the Gigahertz range. Rather, the sampling rate of the sample and hold circuit <b>154</b> can be set much higher than twice the bandwidth of the coarse frequency region of interest <b>170</b><i>c</i>, which allows the passband, transition band, and stopband requirements of the coarse bandpass filter <b>150</b> to be relaxed. Further, the amplitude response of the coarse bandpass filter <b>150</b> in the passband also does not have to be flat since it can be corrected digitally as is further described below with relation to <figref idref="DRAWINGS">FIG. 9</figref>. However, in other embodiments other types of sampling can be used such as direct sampling or over-sampling rather than sub-sampling.

0063The amount of attenuation provided by the coarse bandpass filter <b>150</b> is chosen in relation to the sampling frequency used by the sample and hold circuit <b>154</b>, the bandwidth of the coarse frequency region of interest <b>170</b><i>c</i>, and the amount of resolution required for the ADC <b>106</b>. For example, assume that the sample and hold circuit <b>154</b> uses a sampling rate of 500 MHz, the passband of the coarse bandpass filter <b>150</b> is about 500 MHz, and the mixing stage <b>128</b> mixes the desired television channel signal <b>22</b> to 1.125 GHz. This sub-sampling, otherwise known as undersampling, places an image of the desired television channel signal <b>22</b> that was originally in the 1.125 GHz region at 125 MHz, and other images <b>170</b><i>a </i>and <b>170</b><i>b </i>of the output signal <b>170</b> of the coarse bandpass filter <b>150</b> centered at multiples of 250 MHz away from 125 MHz as shown in <figref idref="DRAWINGS">FIG. 5C</figref>. As can be seen, the attenuation provided by the coarse bandpass filter <b>150</b> is such that the sub-sampled version of its output <b>170</b> overlaps with images <b>170</b><i>a </i>and <b>170</b><i>b</i>. However, the attenuation provided by the coarse bandpass filter <b>150</b> is selected such that at the region of overlap between the sub-sampled version of the coarse frequency region of interest <b>170</b><i>c </i>and the skirts of the images <b>170</b><i>a </i>and <b>170</b><i>b</i>, the skirts of the images <b>170</b><i>a </i>and <b>170</b><i>b </i>are sufficiently attenuated to provide an adequate amount of resolution without aliasing when the ADC <b>106</b> samples the frequency region of interest <b>170</b><i>c</i>. For example, in at least some cases, the amount of attenuation can be at least −74 dB at +/−BW/2 from 125 MHz to ensure that the ADC <b>106</b> will have 12-bit resolution where BW/2 is half of the bandwidth of the coarse bandpass filter <b>150</b>. It should be noted that a smaller bandwidth can be selected for the coarse bandpass filter <b>150</b> if inductors with a higher Q are employed. For example, Q-enhanced inductors, bond-wires, and the like can be used to increase the Q to 5 and reduce the bandwidth to about 250 MHz or so.

0064In some implementations, the coarse bandpass filter <b>150</b> can be realized as a 6^thorder filter (see <figref idref="DRAWINGS">FIG. 6A</figref>) which can be implemented using a cascade of three 2^ndorder LC tank filter stages, similar to the LC tank filters used in the mixing stage <b>128</b>. Each tank filter stage can be separated by a buffer to avoid interaction with one another. Furthermore, each buffer can be used as a distributed source of gain control to help maintain a reasonable signal level as the signal power decreases with each subsequent LC tank filter stage. The overall Q of the coarse bandpass filter <b>150</b> can be about 12. The coarse bandpass filter <b>150</b> can be implemented in other ways, as is commonly known by those skilled in the art.

0065The sample and hold circuit <b>154</b> is provided with a timing signal by the frequency synthesizer <b>160</b> to perform sub-sampling. The frequency synthesizer <b>160</b> receives a timing control signal <b>166</b> from the digital processing block <b>108</b> so that the rate of sub-sampling can be varied if needed. An aperture window is associated with the sample and hold circuit <b>154</b>, and the length of the aperture window can be selected such that the multi-channel television signal <b>136</b> can be resolved to the least significant bit of the ADC <b>106</b>. It should be understood that all blocks following the sample and hold circuit <b>154</b> are implemented with discrete-time components.

0066The discrete-time coarse bandpass filter <b>156</b> can be used to provide another level of filtering. The discrete-time coarse bandpass filter <b>156</b> can be realized as a switched capacitor filter with a passband centered at 125 MHz, for this example. The discrete-time coarse bandpass filter <b>156</b> similarly provides a coarse channel signal output that includes the desired television channel signal <b>22</b>. However, the discrete-time coarse bandpass filter <b>156</b> has a sharper transfer function than the coarse bandpass filter <b>150</b> in that it provides a larger amount of attenuation (i.e. the rolloff is larger) to deal with the more extreme blocking profiles in which the adjacent television channel signal may be 35 to 40 dB larger than the desired television channel signal <b>22</b> (this depends on the television broadcast standard). In some implementations, the discrete-time coarse bandpass filter <b>156</b> can be realized as an 8^thorder filter (see <figref idref="DRAWINGS">FIG. 6B</figref>) with a much steeper rolloff than that of the coarse bandpass filter <b>150</b> to further improve the signal-to-noise ratio (SNR) for the desired television channel signal <b>22</b>. In some cases, the discrete-time coarse bandpass filter <b>156</b> can also be used to limit the power of the coarse channel signal <b>162</b> such that the resolution of the ADC <b>106</b> is sufficient to digitize this signal and to resolve the desired television channel signal <b>22</b> to the necessary accuracy.

0067The VGAs <b>152</b> and <b>158</b> can provide an appropriate amount of gain to the output of the coarse bandpass filters <b>150</b> and <b>156</b>, respectively, based on the amount of filtering that was done by these filters and the level of the received signal. Furthermore, due to the diverse RF signal blocking profiles (one of which is shown in <figref idref="DRAWINGS">FIG. 1B</figref> for example), various gain combinations can be selected to yield improved performance. Accordingly, the VGAs <b>124</b>, <b>134</b>, <b>152</b> and <b>158</b> receive gain control signals <b>138</b>, <b>142</b>, <b>164</b> and <b>168</b> from the digital processing block <b>108</b> to more effectively apply a distributed amount of gain or attenuation.

0068The digital processing block <b>108</b> determines the optimal distribution and the amount of gain/attenuation. The digital processing block <b>108</b> can determine gain distribution in a number of ways. For instance, the digital processing block <b>108</b> can determine gain distribution based on measurements that are made by the power meter <b>126</b>. In this case, the power meter <b>126</b> provides analog signal information <b>144</b> to the digital processing block <b>108</b> (this can be provided in a digital manner). Alternatively, the digital processing block <b>108</b> can use other methods for controlling the gain of the RF and analog processing blocks <b>102</b> and <b>104</b> in which case the power meter <b>126</b> can be optional.

0069Generally, the amount of gain provided by the RF processing block <b>102</b> and the analog processing block <b>104</b> can be adjusted to affect sensitivity and distortion of signals at various locations in the RF and analog processing blocks <b>102</b> and <b>104</b>. The amount of gain provided by the analog processing block <b>104</b> can be adjusted such that the input range of the ADC <b>106</b> is fully utilized while controlling the amount of distortion in blocks <b>102</b> and <b>104</b>. In some cases, the entire amount of gain provided by the RF and analog processing blocks <b>102</b> and <b>104</b> can be on the order of 100 dB. Gain control techniques that can be used are described in more detail below with reference to <figref idref="DRAWINGS">FIGS. 15</figref>, and <b>16</b>A-<b>16</b>C.

0070The ADC <b>106</b> digitizes the output of the analog processing block <b>104</b> to provide a digitized coarse channel signal <b>172</b>. The digitized coarse channel signal <b>172</b> includes the desired television channel signal <b>22</b>, and portions of one or more adjacent television channel signals depending on the television broadcast standard. The coarse filtering provided by the various components in the RF processing block <b>102</b> and the analog processing block <b>104</b> dictate the number of bits that are required for the ADC <b>106</b>. If the mixing stage <b>128</b> and the coarse bandpass filters <b>150</b> and <b>156</b> provide more filtering, then the ADC <b>106</b> can be implemented with a smaller number of bits. Accordingly, selecting the filtering characteristics of these components represents a balance between the complexity of providing greater selectivity in filtering versus the need to provide higher resolution in the ADC <b>106</b>.

0071The ADC <b>106</b> can be realized with a bandpass delta sigma (BDS) ADC that can provide 11 or 12 effective bits (1 bit provides about 6 dB of resolution/gain for the sampled signal). The BDS ADC oversamples at a very high rate with 3-4 actual bits to produce 11 or 12 effective bits for digitization. The BDS ADC outputs the data at an IF frequency. However, in other implementations it may be possible to realize the ADC <b>106</b> with a 12-bit Nyquist rate ADC.

0072In alternative embodiments, the sample and hold circuit <b>154</b> and the coarse bandpass filter <b>156</b> can be realized with a continuous-time filter and the VGA <b>158</b> with a traditional continuous-time VGA if a continuous-time bandpass sigma-delta converter is used for the ADC <b>106</b>. In some cases, a lowpass ADC can be used provided that the appropriate filtering in the analog processing block <b>104</b> precedes the lowpass ADC. The input intermediate frequency can also be altered.

0073Also, in alternative embodiments, depending on the frequency range of the desired television channel signal <b>22</b> as well as any mixing or other frequency shifting that is employed by the RF and analog processing blocks <b>102</b> and <b>104</b>, it is possible to replace at least some of the coarse bandpass filters <b>150</b> and <b>156</b> described for the analog processing block <b>104</b> with coarse lowpass filters.

0074Referring now to <figref idref="DRAWINGS">FIG. 7</figref>, shown therein is a block diagram of an exemplary embodiment of the digital processing block <b>108</b>. The digital processing block <b>108</b> includes an input filtering block <b>180</b>, a video processing block <b>182</b>, first and second audio filtering blocks <b>184</b> and <b>186</b>, and an audio processing block <b>188</b>. The digital processing block <b>108</b> also includes a control block <b>190</b> for controlling various blocks in the digital processing block <b>108</b> as well as providing timing and control signals to various components in the RF and analog processing blocks <b>102</b> and <b>104</b>. Certain portions of the digital processing block <b>108</b> operate according to the sampling rate used by the sample and hold circuit <b>154</b>; however, other sampling rates are also used by employing downsampling or interpolation. The digital processing block <b>108</b> is generally configured to operate in an analog operation mode for processing signals transmitted according to an analog broadcast standard, or a digital processing mode for processing signals transmitted according to a digital broadcast standard.

0075The digital processing block <b>108</b> generally processes the digitized coarse channel signal <b>172</b> to recover the video and audio information for the desired television channel signal <b>22</b>. The processing takes into account the television broadcast standard used to transmit the desired television channel signal <b>22</b>. The digital processing block <b>108</b> can produce various outputs, depending on the particular implementation. These outputs generally include, in different embodiments, various combinations of: digitized versions of a CVBS (Composite Video Baseband Signal) output for PAL/SECAM/NTSC formats, left and right channel audio outputs, a sound IF output that can be further processed by an audio decoder (not shown), and a digital video output, which can be further processed by a digital television demodulator to provide a digital transport stream that can then be processed by another element such as an MPEG-2 decoder, for example. The digital transport stream includes compressed digital data representing the audio and video information of one or more television programs. Additionally, a digital IF output may be provided to an external digital demodulator. The digitized versions of the CVBS, baseband audio, sound IF and digital IF outputs may be converted to analog form using the DAC block <b>110</b>. In alternative embodiments, these signals may be conveyed in digital form to subsequent processing stages without need for further conversion.

0076More particularly, the digitized coarse channel signal <b>172</b> is processed by the input filtering block <b>180</b> to provide a processed digitized coarse channel signal <b>192</b>. The input filtering block <b>180</b> generally provides a combination of down conversion, pre-filtering and downsampling. The input filtering block <b>180</b> also includes components for setting the gain of various variable gain amplifiers in the RF and analog processing blocks <b>102</b> and <b>104</b>. However, in alternative embodiments this functionality can be provided by the control block <b>190</b>. The processed digitized coarse channel signal <b>192</b> is then processed by the video processing block <b>182</b>, which provides output signals <b>194</b> according to a desired output format. The output signals <b>194</b> include only video information if the television broadcast standard is analog. However, if the television broadcast standard is digital, then the output signals <b>194</b> include video and audio content in a format that is modulated according to the broadcast standard. Further, the video processing block <b>182</b> has two modes of operation: an analog operation mode to process the desired television channel signal <b>22</b> when it is transmitted according to an analog broadcast standard, and a digital operation mode to process the desired television channel signal <b>22</b> when it is transmitted according to a digital broadcast standard. This is described in further detail below.

0077If the desired television channel signal <b>22</b> is transmitted according to an analog broadcast standard, then the processed digitized coarse channel signal <b>192</b> is also processed by at least one of the first and second audio filtering blocks <b>184</b> and <b>186</b> which provide at least one of intermediate audio signals <b>196</b> and <b>198</b>. In alternative embodiments, as explained in further detail below with relation to <figref idref="DRAWINGS">FIG. 13B</figref>, instead of receiving the processed digitized coarse channel signal <b>192</b>, the first and second audio filtering blocks <b>184</b> and <b>186</b> can receive a processed version of this signal provided by the video processing block <b>182</b>. The first and second audio filtering blocks <b>184</b> and <b>186</b> can also provide sound IF signals SIF<b>1</b> and SIF<b>2</b>. The intermediate audio signals <b>196</b> and <b>198</b>, and the sound IF signals SIF<b>1</b> and SIF<b>2</b> are then processed by the audio processing block <b>188</b> to provide an audio output signal <b>200</b> according to a desired output format. The first and second audio filtering blocks <b>184</b> and <b>186</b> may both be used for situations in which the analog television broadcast standard dictates the use of two audio carriers. Alternatively, in this case, a single audio separation block can be used to separate both carriers at the same time and to provide them via a single SIF connection to a downstream audio device for further processing. For analog broadcast standards that utilize a single audio carrier, only one of the first and second audio filtering blocks <b>184</b> and <b>186</b> are enabled by the control block <b>190</b>.

0078In at least some cases, the control block <b>190</b> can also receive signals B<b>1</b> and B<b>2</b> from the video processing block <b>182</b> and a digital demodulator respectively to determine whether the television broadcast standard used for the desired television channel signal <b>22</b> is analog or digital. For instance, the video processing block <b>182</b> can be first operated in the analog operation mode and if a lock to the picture carrier is achieved in a reasonable amount of time then this lock is identified via the signal B<b>1</b> (see <figref idref="DRAWINGS">FIG. 12A</figref>) so that the control block <b>190</b> can configure the various components of the digital processing block <b>108</b> for analog operation mode. If a lock is not obtained, then the control block <b>190</b> can configure the video processing block <b>182</b> for operation in the digital operation mode and determine if the desired television channel signal <b>22</b> is properly demodulated. In this regard, a component of a digital demodulator (see <figref idref="DRAWINGS">FIG. 17B</figref>) can indicate successful demodulation in the signal B<b>2</b>.

0079The control block <b>190</b> also determines the particular type of analog or digital television broadcast standard. One way to determine the television broadcast standard is to detect the type of audio information that is included in the processed digitized coarse channel signal <b>192</b>. Alternatively, the universal television receiver <b>100</b> can be configured for a particular broadcast standard. In this respect, the architecture of the universal television receiver <b>100</b> allows the universal television receiver <b>100</b> to be mass-produced and then configured, by setting certain parameters, for processing television signals transmitted according to a certain broadcast standard.

0080Referring now to <figref idref="DRAWINGS">FIG. 8</figref>, shown therein is a flow chart diagram of an exemplary embodiment of a standards detection method <b>250</b> which can be used for detecting the television transmission broadcast standard that was used to transmit the wideband television signal <b>10</b>. The standards detection method <b>250</b> includes a digital detection mode for detecting whether the desired television channel signal <b>22</b> is a digital television signal and an analog detection mode for detecting whether the desired television channel signal <b>22</b> is an analog television signal. This includes receiving feedback from the video and audio processing blocks <b>182</b>-<b>188</b>. Once the bandwidth of the desired television channel signal <b>22</b> is found, the control block <b>190</b> sets the bandwidth (BW) and the mode of operation for the digital processing block <b>108</b>.

0081At step <b>252</b>, the standards detection method <b>250</b> enters digital detection mode to determine whether the desired television channel signal <b>22</b> was transmitted with a digital broadcast standard. A first digital broadcast standard is selected and the method <b>250</b> attempts to demodulate and decode the desired television channel signal <b>22</b> according to the selected digital broadcast standard using methods known by those skilled in the art. If a lock is obtained, then the method <b>250</b> moves to step <b>256</b> and sets the bandwidth used in various blocks in the video processing block <b>182</b> during digital reception mode. If a lock is not obtained, the method <b>250</b> moves to step <b>258</b> to determine whether there are any other digital broadcast standards to check. If so, another digital broadcast standard is checked and the method <b>250</b> moves back to step <b>254</b>. If there are no more digital broadcast standards to check, the method <b>250</b> moves to step <b>260</b>.

0082At step <b>260</b>, the method <b>250</b> enters analog detection mode. At step <b>262</b>, the method <b>250</b> attempts to lock to a carrier frequency. If a carrier frequency is locked, the method <b>250</b> must confirm that the carrier frequency is a picture carrier frequency and not an audio carrier. Several techniques can be used to lock to a carrier frequency and confirm that it is a picture carrier. One technique includes using a coarse version of sync decoding to determine if sync information is associated with the locked carrier frequency. If a lock has been made to a picture carrier frequency, the method <b>250</b> then moves to 264 otherwise the method <b>250</b> continues to attempt to lock to a picture carrier frequency. The coarse sync detection method involves counting the lines per field to verify that the locked carrier is a picture carrier.

0083At step <b>264</b>, the method <b>250</b> attempts to locate the audio carrier. To accomplish this, a look-up table may be consulted which includes the possible positions of the audio carrier based on the analog broadcast transmission standard. The method <b>250</b> can start with the audio carrier that is positioned furthest away from the picture carrier frequency to determine if an audio carrier exists at that position. If a signal is found at the audio carrier frequency then it is assumed to be an audio signal but can be checked to make sure that it is not another picture carrier by performing the coarse sync decoding. If an audio carrier is not located, then the method <b>250</b> looks at the audio carrier that is second furthest from the located picture carrier and repeats this process iteratively until detecting the audio carrier. If the audio carrier is detected, the method <b>250</b> can search for a second audio carrier since some analog broadcast standards employ two audio carriers. Once the single or dual audio carriers are detected, the method <b>250</b> moves to step <b>256</b> to set the BW and the mode of operation for the digital processing block <b>108</b>. The BW can be set to be the difference between the picture carrier frequency and the audio carrier frequency minus a few hundred KHz, so as to ensure that the audio signal is adequately attenuated in the video path.

0084If the audio carrier is not located at step <b>264</b>, then the method <b>250</b> moves to step <b>266</b> to search for the next picture carrier signal. The search range for the next picture carrier signal can be the widest bandwidth and guard band that is used in analog broadcast transmission standards. If no picture carrier signal is located, then the BW can be set to the maximum bandwidth for analog broadcast transmission standards. If another picture carrier is found, then the bandwidth can be determined based on the distance of the newly located picture carrier frequency compared to the initially located picture carrier frequency.

0085Based on the detected television broadcast standard for the desired television channel signal <b>22</b>, the control block <b>190</b> provides a video control information signal <b>202</b> to the video processing block <b>182</b> to control the operation mode of the video processing block <b>182</b>. The control block <b>190</b> also provides audio control information signals <b>204</b>, <b>206</b> and <b>208</b> to the first and second audio filtering blocks <b>184</b> and <b>186</b>, and the audio processing block <b>188</b>. Generally, the audio control information signals <b>204</b>, <b>206</b> and <b>208</b> enable blocks <b>184</b>, <b>186</b> and <b>188</b> when the television broadcast standard used for the desired television channel signal <b>22</b> is analog. In this case, the audio control information signals <b>204</b> and <b>206</b> enable at least one of the audio filtering blocks <b>184</b> and <b>186</b> depending on the number of audio carriers in the processed digitized coarse channel signal <b>192</b>. The audio control information signals <b>204</b>, <b>206</b> and <b>208</b> can include other operational parameter values that are discussed in more detail below. The control block <b>190</b> also provides control and timing information signal <b>210</b> to the RF and analog processing blocks <b>102</b> and <b>104</b>. This information is derived from input information <b>212</b> based on the television channel that a user of the television receiver <b>100</b> wishes to view. The control and timing information signal <b>210</b> is used to control the frequencies of various synthesizers as well as sampling rates.

0086Referring now to <figref idref="DRAWINGS">FIG. 9</figref>, shown therein is a block diagram of an exemplary embodiment of the input filtering block <b>180</b>. The input filtering block <b>180</b> includes a frequency rotator (i.e. a down-converter or digital mixer) <b>300</b> which receives a rotation control signal <b>302</b>, a decimation filtering block <b>304</b>, an equalizer <b>306</b>, and an analog gain control block <b>308</b>. It should be noted that in alternative embodiments, the analog gain control block <b>308</b> can be located in a different block. Further in some alternative embodiments, the functionality of the analog gain control block <b>308</b> can be provided by the control block <b>190</b>.

0087The frequency rotator <b>300</b> processes the digitized coarse channel signal <b>172</b> by performing down conversion to the baseband such that the coarse frequency region of interest <b>170</b><i>c </i>is now centered about DC. The amount of down conversion is controlled by the rotation control signal <b>302</b>, which is provided by a free-running discrete time oscillator (not shown). The choice of a sampling rate that is 4 times the centre frequency of the range of interest allows the rotator <b>300</b> to be implemented without the need for multipliers, since rotation occurs in 90 degree increments. The frequency rotator <b>300</b> generates quadrature signals, i.e. in-phase (I) and quadrature (Q) signals for improved processing efficiency. The I and Q signals are then processed by separate I and Q signal paths as is commonly known by those skilled in the art. It should be noted that only one signal path with double/thicker lines are shown to denote the I and Q signal paths to simplify the description. However, it should be understood that blocks with two I and Q inputs and two I and Q outputs are actually implemented with two blocks; one block processes the I signal and the other block processes the Q signal. This nomenclature is used in other figures as well.

0088The decimation filtering block <b>304</b> provides some low pass filtering and downsampling to the output of the frequency rotator <b>300</b> to remove unwanted signal components and to reduce the sample rate in order to simplify subsequent processing stages. The filtering removes quantization noise that results from the digitization provided by the ADC <b>106</b>. If a bandpass sigma delta converter is used for the ADC <b>106</b>, then the filtering can be designed to attenuate the noise-shaped spectral regions of the I and Q output signals of the frequency rotator <b>300</b>. The decimation filtering block <b>304</b> then performs downsampling so that the other blocks in the digital processing block <b>108</b> can operate more efficiently. For example, downsampling can be done so that the sampling rate associated with the processed digitized coarse channel signal <b>192</b> is at approximately 31.25 MHz. The amount of low pass filtering provided by the decimation filtering block <b>304</b> is also commensurate with the amount of downs ampling as is commonly known by those skilled in the art.

0089The output of the decimation filtering block <b>304</b> is then processed by the equalizer <b>306</b>. It should be noted that the equalizer <b>306</b> is optional and is not needed if the equalization functionality can be provided by another downstream element; for example, equalization can be provided by a digital demodulator (not shown) that receives the output signal <b>194</b> of the video processing block <b>182</b>. The functionality of the equalizer <b>306</b> may also be optional when processing television signals that are transmitted according to an analog broadcast standard, depending on the level of performance required for the output signals <b>194</b>. Accordingly, in some cases the equalizer <b>306</b> can be disabled or not included.

0090The equalizer <b>306</b> processes the I and Q output of the downs ampler to compensate for the non-ideal filtering provided by the coarse bandpass filters <b>150</b> and <b>156</b> in the analog processing block <b>104</b>. Accordingly, the equalizer <b>306</b> provides equalization to make it seem as if the coarse bandpass filters <b>150</b> and <b>156</b> have a flat inband response and improved group delay response. The equalizer <b>306</b> can combine the I and Q signals to provide a single-ended real output signal <b>192</b>. Alternatively, in some alternative embodiments, the equalizer <b>306</b> can provide I and Q output signals that are then appropriately processed by the other blocks in the digital processing block <b>108</b>.

0091The equalizer <b>306</b> can be implemented in a variety of fashions as is commonly known by those skilled in the art. For instance, the equalizer <b>306</b> can include a real or imaginary FIR filter to perform equalization, and can also include several cascaded filters as well as a frequency rotator to shift the signal by a certain fraction of the sampling rate so that the filters in the equalizer <b>306</b> can be implemented more efficiently. In this case, the blocks <b>182</b>, <b>184</b> and <b>186</b> include a corresponding frequency rotator to shift the processed digitized coarse channel signal <b>192</b> back to the baseband.

0092The output of the ADC <b>106</b> is also provided to the analog gain control block <b>308</b>. The purpose of the analog gain control block <b>308</b> is to adjust the gain settings at the RF and IF stages to adjust the level of the signal presented to the ADC <b>106</b> to ultimately increase the quality of the desired television channel signal <b>22</b>. This may be performed using a feedback loop in which the level of the output from the ADC <b>106</b> is measured and compared to a preset reference level. If the measured level is less than the reference level then the gain is increased, while if the measured level is higher than the reference level then the gain is decreased. Therefore, the reference level may be thought of as a target level which the loop seeks to maintain, even when the level of the signal from the antenna <b>120</b> changes. The analog gain control block <b>308</b> can perform this gain control technique, which is described in more detail with regards to <figref idref="DRAWINGS">FIGS. 16A-16C</figref>.

0093The analog gain control block <b>308</b> is implemented digitally and provides RF gain control signals <b>138</b> and <b>142</b> to the VGAs <b>124</b> and <b>134</b>, respectively, in the RF processing block <b>102</b>. The analog gain control block <b>308</b> also provides IF gain control signals <b>164</b> and <b>168</b> to the VGAs <b>152</b> and <b>158</b>, respectively, in the analog processing block <b>104</b> if these amplifiers exist (note they are optional). The analog gain control block <b>308</b> can perform the gain control method outlined in <figref idref="DRAWINGS">FIG. 15</figref> or other methods described in further detail below. In alternative embodiments, a gain control system may be implemented which utilizes information provided by the power meter <b>126</b>, in which case the analog gain control block <b>308</b> also receives the analog signal information <b>144</b>.

0094Referring now to <figref idref="DRAWINGS">FIG. 10</figref>, shown therein is a block diagram of an exemplary embodiment of the video processing block <b>182</b>. The video processing block <b>182</b> processes the processed digitized coarse channel signal <b>192</b> by generally performing carrier frequency recovery, resampling and filtering. The video processing block <b>182</b> can also perform phase noise reduction by compensating for phase perturbations which includes both noise and systematic variation caused by spurs and the like. As mentioned, the video processing block <b>182</b> operates in a digital operation mode or an analog operation mode depending on whether the desired television channel signal <b>22</b> is transmitted according to a digital or analog television broadcast standard.

0095The video processing block <b>182</b> includes a first frequency rotator <b>350</b>, a video pre-polyphase filter <b>352</b><i>p</i>, a first video polyphase filter <b>352</b>, a video resampling control block <b>354</b>, a video filter <b>356</b>, a digital VGA <b>358</b>, a multiplexer <b>360</b>, a digital gain control block <b>362</b>, a second frequency rotator <b>364</b>, a picture carrier recovery block <b>366</b>, an output equalizer <b>388</b>, a second video polyphase filter <b>368</b>, an up-sampling block <b>370</b> and a Digital to Analog Converter (DAC) <b>372</b>. The video pre-polyphase filter <b>352</b><i>p</i>, first video polyphase filter <b>352</b>, video resampling control block <b>354</b>, and the video filter <b>356</b> can be considered to be a video filter stage. The picture carrier recovery block <b>366</b> includes a carrier recovery filter <b>374</b>, a first phase rotator <b>376</b>, a carrier recovery block <b>378</b>, an AGC filter <b>380</b>, a second phase rotator <b>382</b>, a Vestigial Side Band (VSB) filter <b>384</b>, a third phase rotator <b>386</b>, an overmodulation filter <b>406</b>, and an overmodulation magnitude detector <b>408</b>.

0096The control block <b>190</b> provides a mode control signal <b>389</b> to control whether the video processing block <b>182</b> operates in the analog or digital operation mode. The mode control signal <b>389</b> is provided to the multiplexer <b>360</b> as a selection input to select which gain control signal is applied to the digital VGA <b>358</b>. This is described in further detail below. The mode control signal <b>389</b> is also provided to the picture carrier recovery block <b>366</b> to enable this block during the analog operation mode or to disable this block during the digital operation mode.

0097In both digital and analog operation modes, the frequency rotator <b>350</b> generally shifts the frequency content of the processed digitized coarse channel signal <b>192</b> so that the desired television channel is centered about DC. In the case of analog reception, the video information including the vestigial sideband of the desired channel is centered about DC. In the case of digital reception, the entire channel is similarly centered. Both of these cases are generally shown in <figref idref="DRAWINGS">FIG. 11A</figref>. The frequency rotator <b>350</b> also produces in-phase I and quadrature Q signals, and the majority of the video processing block <b>182</b> includes two signal paths for processing the I and Q signals. However, there is a degree of offset in the location of the frequency content of the desired television channel due to both known and unknown elements. The degree of offset can be larger for the analog operation mode compared to the digital operation mode for some situations. A portion of the offset results from coarse positioning of the frequency content of the desired television channel signal due to using a coarse step size or a fine step size in the frequency synthesizer <b>132</b> and this amount of offset is known. However, an additional offset arises from frequency tolerances in the frequency synthesizer <b>132</b>, and the transmitter, or other hardware, that transmitted the desired television channel signal <b>22</b> to the universal television receiver <b>100</b>. These offsets are unknown and must also be corrected.

0098In the digital operation mode, the offset correction can be performed at a later stage by another component such as a digital demodulator (not shown). Those skilled in the art are familiar with techniques that can be employed in the digital demodulator for correcting the offset for television channel signals that are broadcast according to a digital broadcast standard. Accordingly, in some embodiments, there is no feedback signal that is provided to the frequency rotator <b>350</b> during digital operation mode. Rather, the video processing block <b>182</b> can operate in an open-loop or free-running fashion where a fixed or known frequency shift is applied to the processed digitized coarse channel signal <b>192</b> to shift the signal to the baseband to attempt to center the coarse frequency region of interest <b>170</b><i>c </i>about DC in spite of the offset. Accordingly, known frequency offset errors in the carrier frequency, such as errors due to the use of a coarse step size or fine step size in the oscillator used in the RF processing block <b>102</b> or due to any other known frequency offset errors, may be compensated in this way by applying a corresponding known frequency shift via the frequency rotator <b>350</b>. The correction of other offsets that may be unknown can be performed by a downstream digital demodulator (not shown). Alternatively, in other embodiments that include a digital demodulator (see <figref idref="DRAWINGS">FIGS. 17A and 17B</figref> for example), the digital demodulator can provide a feedback signal to the frequency rotator <b>350</b> to adjust the shift that is applied to the processed digitized coarse channel signal <b>192</b> so that it is centered about DC regardless of the offset.

0099In the analog operation mode, the portion of the analog broadcast television signal from frequency f₁to f₂should be centered about DC by the frequency rotator <b>350</b>. For example, <figref idref="DRAWINGS">FIG. 11B</figref> shows the location of frequencies f₁and f₂for an NTSC analog television channel signal as is used in North America. This is achieved by using a feedback loop, which includes the first and second frequency rotators <b>350</b> and <b>364</b> and the picture carrier recovery block <b>366</b>. The frequency rotator <b>350</b> applies a variable frequency shift to the processed digitized coarse channel signal <b>192</b> to attempt to center the frequency content of the video information of the desired television channel signal <b>22</b> about DC. Upon initial operation, the frequency rotator <b>350</b> applies an initial frequency shift. The selection of the initial frequency shift is discussed in more detail with reference to <figref idref="DRAWINGS">FIG. 12A</figref>. The frequency rotator <b>364</b> applies a fixed frequency shift of Δω such that the picture carrier is shifted to DC when the frequency range f₁to f₂is centered about DC at the output of frequency rotator <b>350</b>. Note that initially the picture carrier may not be shifted exactly to DC due to the offset error. The picture carrier recovery block <b>366</b> then tracks the frequency offset error by detecting the actual location of the picture carrier signal and generates an analog mode frequency shift feedback signal <b>390</b>, which is provided to the frequency rotator <b>350</b> to adjust the amount of frequency shift that it provides. Over time, this adjustment in frequency shift is such that the fixed frequency shift provided by the frequency rotator <b>364</b> moves the picture carrier signal to DC. The operation of the picture carrier recovery block <b>366</b> is described in further detail below.

0100The output of the frequency rotator <b>350</b> is processed by the video pre-polyphase filter <b>352</b><i>p </i>which provides low pass filtering to remove unwanted spectral components. The output of the video pre-polyphase filter <b>352</b><i>p </i>is processed by the video polyphase filter <b>352</b> which provides interpolation to change the number of data samples for the output of the video pre-polyphase filter <b>352</b><i>p</i>. The interpolation operation performed by the video polyphase filter <b>352</b> provides a different sampled version of the output of the video pre-polyphase filter <b>352</b><i>p </i>in that the data samples of the output of the video polyphase filter <b>352</b> are at different temporal locations and have a different temporal spacing compared to the output of the video pre-polyphase filter <b>352</b><i>p</i>. The end effect of the video polyphase filter <b>352</b> is to change the sampling rate at the output of the video polyphase filter <b>352</b> such that the bandwidth of the desired television channel signal <b>22</b> normalized with respect to the new sampling rate is transformed to match the bandwidth of the video filter <b>356</b>. Accordingly, the video polyphase filters <b>352</b> and <b>368</b> act as resampling elements. The amount of interpolation (i.e. the amount of resampling) is dictated by the video resampling control block <b>354</b>, which provides resampling control signals <b>392</b> and <b>394</b> to the video polyphase filters <b>352</b> and <b>368</b> respectively. The degree of interpolation that is required is related to the broadcast transmission standard that was determined by the control block <b>190</b>, which is indicated by a broadcast information signal that is derived from the video control information <b>202</b> that is provided by the control block <b>190</b>. In alternative embodiments, the functionality of the video resampling control block <b>354</b> can be provided by the control block <b>190</b>.

0101Consider the parameter W_null=f_null/f_S^{(polyphase output)}which is the fixed cut-off frequency of the video filter <b>356</b> normalized with respect to the new sampling rate f_S^{(polyphase output)}. The video resampling control block <b>354</b> configures the value for the new sampling rate f_S^{(polyphase output)}such that f_nullcorresponds to half of the bandwidth of the desired television channel signal <b>22</b>, since it is a complex signal centered about DC. The value for the parameter f_nullvaries based on the television broadcast standard. For analog television broadcast standards, the desired television channel signal <b>22</b> includes the video information f_vabove the picture carrier frequency as defined by the particular broadcast standard, as well as the vestigial sideband portion f_VSBlocated below the picture carrier frequency both in MHz and shown in <figref idref="DRAWINGS">FIG. 11B</figref>. Accordingly, f_nullcan be given by the following formula. <br /><i>f</i>_null˜(<i>f</i>_v<i>+f</i>_VSB)/2 MHz

0102The value of f_VSBis assumed to be 0.75 MHz in the following examples, though other values may also be possible. The value of W_nullcan be set to 0.31 by design, for example, as a filter with this characteristic represents a practical trade-off in design considerations. Some examples are shown below for various analog television broadcast standards. <br /><i>f</i>_null˜(4.2+0.75)/2 MHz (NTSC)<br /><i>f</i>_null˜(5.0+0.75)/2 MHz (PAL <i>B,G</i>)<br /><i>f</i>_null˜(5.5+0.75)/2 MHz (PAL <i>I</i>)<br /><i>f</i>_null˜(5.7+0.75)/2 MHz (PAL <i>D,K</i>)<br /> For digital television broadcast standards including DVB-T, the bandwidth of the desired television channel signal <b>22</b> is the entire channel width as illustrated by the following examples (f_VSBis not applicable). As in the previous examples, f_nullis taken as half of this bandwidth since the signal is complex in nature. <br /><i>f</i>_null˜(6.0)/2 MHz (6 MHz DVB-T)<br /><i>f</i>_null˜(7.0)/2 MHz (7 MHz DVB-T)<br /><i>f</i>_null˜(8.0)/2 MHz (8 MHz DVB-T)<br /><i>f</i>_null˜(6.0)/2 MHz (6 MHz ATSC)

0103Accordingly, rather than implementing the video filter <b>356</b> as a variable bandpass filter, the video processing block <b>182</b> employs a fixed bandwidth for the video filter <b>356</b> and changes the effective sampling rate of the data provided to the video filter <b>356</b>. The output of the video polyphase filter <b>352</b> is at the same physical clock rate but at a new effective sampling rate to adjust the spectrum or bandwidth of the desired television channel signal <b>22</b> to match the passband of the video filter <b>356</b>. This processing allows a sharp, fixed filtering block to be used as if it has a variable passband size. This results in a more efficient implementation since coefficients for a variety of different filter transfer functions to match each of the television broadcast standards is not needed. Rather just one set of coefficients for the video filter <b>356</b> are stored. This “resampling processing” allows the video processing block <b>182</b> to filter television channel signals having different bandwidths, such as 6, 7 or 8 MHz, with the same fixed filter. Further, it should be noted that the filtering provided by the video filter <b>356</b>, the equalizer <b>306</b>, and the analog filters <b>150</b> and <b>156</b> approximate the filtering provided by the SAW filter that is conventionally used in traditional television receivers. Decimation filtering can also be used prior to the video filter <b>356</b> to reduce the number of data samples, and accordingly the number of coefficients used for the video filter <b>356</b>.

0104The same filter coefficients may be used by the video filter <b>356</b> in the digital and analog operation modes. In the analog reception mode, the sample rate is adjusted such that the video filter <b>356</b> separates the video information of the desired television channel signal <b>22</b> including the vestigial sideband portion, as described earlier. In the digital reception mode, the sample rate is adjusted such that the video filter <b>356</b> passes the entire digital channel, since the video and audio information are transmitted together as a multiplexed data stream. <figref idref="DRAWINGS">FIG. 11C</figref> shows an exemplary diagram for a desired television channel conforming to the ATSC digital broadcast standard as is used in North America. In this case, the effective bandwidth may be set slightly wider than the bandwidth of the desired television channel, in order that some offset in the actual frequency of the received desired television channel may be tolerated without the video filter <b>356</b> impinging on the band edges of the desired television channel. Although a slightly wider filter bandwidth may help in some cases, it may harm performance if a strong adjacent channel is present. The narrow filter option may be used without undesired effects in at least some embodiments by applying frequency offset feedback from the digital demodulator to the frequency rotator <b>350</b> in order to keep the effective bandwidth of the video filter <b>356</b> aligned with the desired television channel. The video filter <b>356</b> can be implemented with a pipeline of smaller filters to make the filtering process more efficient, as is commonly known by those skilled in the art.

0105The operation of the video pre-polyphase filter <b>352</b><i>p</i>, video polyphase filter <b>352</b> and the video filter <b>356</b> will now be described in greater detail with reference to <figref idref="DRAWINGS">FIGS. 11D-11L</figref> (the frequency domain is represented in actual frequencies (Hz) not in normalized frequencies in these figures). The output of the frequency rotator <b>350</b> is first processed by the video pre-polyphase filter <b>352</b><i>p</i>. As shown in <figref idref="DRAWINGS">FIG. 11D</figref>, the output of the frequency rotator <b>350</b> includes the desired television channel signal <b>22</b> and out of band components. The video pre-polyphase filter <b>352</b><i>p </i>removes substantially all of the out-of-band and out-of-interest spectral components that would otherwise alias to the desired television channel signal band after resampling by the video polyphase filter <b>352</b>. Accordingly, the video pre-polyphase filter <b>352</b><i>p </i>attenuates the high frequency components of the output of the frequency rotator <b>350</b>. The frequency response of the pre-polyphase filter <b>352</b><i>p </i>is shown in <figref idref="DRAWINGS">FIG. 11D</figref> by the dashed line. The response is flat within the band of the desired television channel signal <b>22</b>. The stopband of the video pre-polyphase filter <b>352</b><i>p </i>specifies the subsampling ratio of the video polyphase filter <b>352</b>. When the spectral components at the input of the video pre-polyphase filter <b>352</b><i>p </i>are greater than half of the sampling rate at the output of the video polyphase filter <b>352</b>, those components are aliased into the output of the video polyphase filter <b>352</b>. Accordingly, the minimum possible sampling rate at the output of the polyphase filter <b>352</b> is limited by the need to avoid this aliasing. These possible aliased components are shown by the dotted lines for the boundary case (i.e. the minimum possible sampling rate after polyphase sampling f_S^{(polyphase output min)}).

0106The video polyphase filter <b>352</b> reduces the rate f_S^{(polyphase input)}of the input data samples to the output sampling rate f_S^{(polyphase output)}. The resampling ratio is f_S^{(polyphase output)}/f_S^{(polyphase input)}. The video polyphase filter <b>352</b> is able to change the sampling rate within a particular range as limited by a stopband boundary frequency polyphase filter) F_boundary^{(pre-polyphase filter)}of the video pre-polyphase filter <b>352</b><i>p </i>and an upper boundary frequency of the desired television channel signal F_boundary^{(desired television channel signal)}as follows. <br /><i>f</i>_S^{(polyphase output)}<i>−F</i>_boundary^{(pre-polyphase filter)}<i>>F</i>_boundary^{(desired television channel signal)}<br /> The spectrum after the video polyphase filter <b>352</b> in actual frequencies (Hz) is shown in <figref idref="DRAWINGS">FIG. 11E</figref>. Now the bandwidth of the desired television channel signal matches the bandwidth of the main video filter <b>356</b> and the desired television channel signal is not damaged by any aliased signals. Other television channel signals and out of band aliased spectral components are at least partially attenuated by the video pre-polyphase filter <b>352</b><i>p. </i>

0107The output of the video polyphase filter <b>352</b> is processed by the video filter <b>356</b>. The purpose of this arrangement is that by changing the sampling rate, the effective bandwidth of the video filter <b>356</b> may be changed even though the filter itself remains fixed. Video filter <b>356</b> is a non-variable low pass filter with a nearly rectangular frequency response that has a very sharp transition between the passband and the stopband. The frequency response and bandwidth are constant in the normalized frequency domain f/f_S, but in the absolute frequency domain they are effectively varied by changing the sampling rate. The output sampling rate f_S^{(polyphase output)}is chosen such that the effective passband of the video filter <b>356</b> matches the bandwidth of the desired television channel signal <b>22</b> and such that any undesired spectral components are removed. For different television broadcast standards the desired television channel signal <b>22</b> may have a different bandwidth. The operation of the frequency rotator <b>350</b> centers the desired television channel signal <b>22</b> about DC and by controlling the output sampling rate f_S^{(polyphase output)}the effective bandwidth of video filter <b>356</b> is modified to match the bandwidth of the desired television channel signal <b>22</b>.

0108The video filter <b>356</b> filters the output of the video polyphase filter <b>352</b> to pass the frequency content of the desired television channel signal <b>22</b> while rejecting the portions of the adjacent television signals that were in the coarse frequency region of interest <b>170</b><i>c</i>. The video filter <b>356</b> extracts the desired television channel signal <b>22</b> with high precision. The video filter <b>356</b> is a steep rectangular filter that is constant in a normalized frequency domain with respect to sampling rate. The frequency response of the video filter <b>356</b> is shown in <figref idref="DRAWINGS">FIGS. 11E and 11F</figref> and the output of the video filter <b>356</b> is shown in <figref idref="DRAWINGS">FIG. 11F</figref>.

0109The digital VGA <b>358</b> then amplifies the output of the video filter <b>356</b>. In the analog operation mode, the amount of digital gain provided by the digital VGA <b>358</b> is dictated by the digital gain control block <b>362</b>, which provides a digital gain control signal <b>398</b> to the digital VGA <b>358</b>. In the digital operation mode, the amount of gain provided by the digital VGA <b>358</b> is dictated by a downstream digital demodulator block (see <figref idref="DRAWINGS">FIGS. 19A and 19B</figref> for an example), which provides a digital gain control signal <b>400</b> to the digital VGA <b>358</b>. The mode control signal <b>389</b> is provided to the multiplexer <b>360</b> to select the digital gain control signal <b>398</b> if the operation mode is the analog operation mode, and the digital gain control signal <b>400</b> if the operation mode is the digital operation mode. The generation of these signals is described in further detail below.

0110In the digital operation mode, the I and Q output signals from the digital VGA <b>358</b> are provided as the output signal <b>194</b> of the video processing block <b>182</b>. These I and Q signals can then be further processed as desired, for instance, by a downstream digital demodulator, or by other processing elements as is well known by those skilled in the art. The digital demodulator would provide a digital transport stream that can then be operated on by another element such as an MPEG-2 decoder to produce video.

0111In the analog operation mode, the I and Q outputs of the digital VGA <b>358</b> are provided to the frequency rotator <b>364</b>, which provides a frequency shift so that the picture carrier signal for the video component of the desired television channel signal <b>22</b> is shifted to DC. To accomplish this, the frequency rotator <b>364</b> provides a fixed frequency shift Δω that is equal to the difference between the center frequency of the video filter <b>356</b> and the expected location of the picture carrier frequency.

0112The I and Q output signals of the frequency rotator <b>364</b> are provided to the picture carrier recovery block <b>366</b> to determine a value for the analog mode frequency shift feedback signal <b>390</b> so that the frequency rotator <b>350</b> can center the frequency content of the desired television channel signal <b>22</b> about DC. The control block <b>190</b> enables the picture carrier recovery block <b>366</b> to operate in the analog operation mode via the mode control signal <b>389</b>. The picture carrier recovery block <b>366</b> uses the picture carrier frequency to demodulate the analog video information in the output of the frequency rotator <b>364</b> to provide a CVBS output as the output signals <b>194</b>.

0113More particularly, the output of the frequency rotator <b>364</b> is provided to the carrier recovery filter <b>374</b>, the AGC filter <b>380</b>, the VSB filter <b>384</b>, and the overmodulation filter <b>406</b>. In alternative embodiments, these blocks can be implemented in one block for improved efficiency by reusing elements required for filtering such as registers. Furthermore, in some cases, some of the filters <b>374</b>, <b>380</b> and <b>406</b> can be implemented using the same filter coefficients as described below.

0114The carrier recovery filter <b>374</b> is a narrowband filter centered at DC with a bandwidth that is wide enough to pass only the picture carrier signal <b>22</b> in order to separate it from the rest of the desired television channel signal <b>22</b> (see <figref idref="DRAWINGS">FIG. 11B</figref>). The carrier recovery filter <b>374</b> filters its input to produce a filtered picture carrier signal. The phase of the filtered picture carrier signal is then rotated by the phase rotator <b>376</b> to compensate for phase noise in the output of the frequency rotator <b>364</b>. The phase rotator <b>376</b> applies a phase adjustment to produce a phase-adjusted filtered picture carrier signal that is then provided to the carrier recovery block <b>378</b>. Phase noise and the operation of the carrier recovery block <b>378</b> is described in more detail with relation to <figref idref="DRAWINGS">FIG. 12A</figref>. In alternative embodiments, the control block <b>190</b> can provide a bandwidth control signal (not shown) to control the bandwidth of this narrowband filter. The bandwidth can be controlled during different stages of the picture carrier recovery process such as the acquisition stage versus the lock stage; this is discussed in more detail with reference to <figref idref="DRAWINGS">FIG. 12A</figref>.

0115The AGC filter <b>380</b> is employed to precondition the output of the frequency rotator <b>364</b> which is a frequency-shifted version of the desired television channel signal <b>22</b>. The AGC filter <b>380</b> performs preconditioning to remove noise and spurious signals. In some cases, the AGC filter <b>380</b> can have the same filter coefficients as the carrier recovery filter <b>374</b>. The AGC filter <b>380</b> filters its input to produce another filtered picture carrier signal, the phase of which is then rotated by the phase rotator <b>382</b> to compensate for phase noise in the output of the frequency rotator <b>364</b>. Accordingly, the phase rotator <b>382</b> applies a phase adjustment to produce another phase-adjusted filtered picture carrier signal, which is provided to the digital gain control block <b>362</b>. Thus, during the sync pulses, when the desired television channel signal <b>22</b> is at a peak level, the digital gain control block <b>362</b> can react to the peak level of the desired television channel signal <b>22</b> rather than noise or signal spikes so that the digital gain control block <b>362</b> can properly adjust the amount of gain applied to the digital VGA <b>358</b> via the digital gain control signal <b>398</b>. Accordingly, the digital gain control block <b>362</b> can include a sync detector to detect the sync pulses and adjust the value of the digital gain control signal <b>398</b>. In other embodiments, the phase rotator <b>382</b> can be eliminated when the digital gain control block <b>362</b> computes the magnitude of the output of the AGC filter <b>380</b> since the magnitude function is phase insensitive.

0116In addition, the gain provided by the digital gain control block <b>362</b> can be adjusted on a line-by-line basis by detecting the magnitude of each sync pulse. Furthermore, to determine the amount of gain that should be applied by the digital VGA <b>358</b>, a desired target level can be set. The difference between the desired target level and the detected peak level of the desired television channel signal <b>22</b> can be used to determine the value of the digital gain control signal <b>398</b> so that the difference tends to zero.

0117The overmodulation filter <b>406</b> is employed to remove noise and spurious signals at the output of the frequency rotator <b>364</b>. The overmodulation magnitude detector <b>408</b> then detects the magnitude of the I and Q components of the output of the overmodulation filter <b>406</b> and provides a magnitude level signal <b>402</b> to the carrier recovery block <b>378</b>, the purpose of which is described further below in relation to <figref idref="DRAWINGS">FIG. 12A</figref>.

0118The video signal from the output of the frequency rotator <b>364</b> is processed by the VSB filter <b>384</b>. The VSB filter <b>384</b> filters the output of the frequency rotator <b>364</b> (i.e. the frequency-shifted version of the desired television channel signal <b>22</b>) to produce filtered video information, i.e. the video information of the desired television channel signal <b>22</b>. The output of the VSB filter <b>384</b> is processed by the phase rotator <b>386</b> and the applied phase correction value is calculated by carrier recovery block <b>378</b>. The VSB filter <b>384</b> and the phase rotator <b>386</b> together convert the input complex vestigial sideband signal to a real CVBS video signal. Under the condition that carrier recovery lock has been acquired, i.e. at the output of the phase rotator <b>386</b>, the picture carrier is shifted to DC and its phase is 0 so that it is completely a real signal, the CVBS signal can be obtained by inverting the sign of the imaginary part spectral components that have negative frequencies, leaving the positive frequency components unchanged and adding both to the real part of the complex input signal. The spectrum of the complex input signal is shown in <figref idref="DRAWINGS">FIGS. 11G to 11I</figref> while the frequency response of the VSB filter <b>384</b> is shown in <figref idref="DRAWINGS">FIGS. 11J to 11L</figref>. Since the final phase rotation by the phase rotator <b>386</b> is done after the complex VSB filter <b>384</b>, the VSB filter <b>384</b> has to generate real and imaginary parts that are provided as an input to the phase rotator <b>386</b>. The CVBS signal is the real output of the phase rotator <b>386</b>. To generate the real output, the VSB filter <b>384</b> multiplies the imaginary part of the input signal spectrum S_VS^Im(f) by the imaginary part of the frequency response H_VSB^Im(f) of the VSB filter <b>384</b>, the product of which is added to the product of the real part of the input signal S_VS^Re(f) and the real part of the frequency response H_VSB^Re(f) of the VSB filter <b>384</b>. It should be noted that the real part of the frequency response H_VSB^Re(f) of the VSB filter <b>384</b> is unity at all frequencies. To generate the imaginary output, the VSB filter <b>384</b> multiplies the real part of the input signal spectrum S_VS^Re(f) by the imaginary part of the frequency response H_VSB^Im(f) of the VSB filter <b>384</b>, the product of which is added to the product of the imaginary part of the input signal S_VS^Im(f) and the real part of the frequency response H_VSB^Re(f) of the VSB filter <b>384</b>. The phase rotator <b>386</b> then applies a phase adjustment to the phase of the output of the VSB filter <b>384</b> to produce phase-adjusted video information which has been compensated for the phase noise in the output of the frequency rotator <b>364</b>. The “real-only” output of the phase rotator <b>386</b> is then provided to the output equalizer <b>388</b>, which applies group delay correction according to television broadcast standards for the desired television channel signal <b>22</b>. The output of the output equalizer <b>388</b> is then provided to the video polyphase filter <b>368</b> so that the signal can be re-interpolated to correspond with the original sampling rate. The output of the video polyphase filter <b>368</b> is then up-sampled by the up-sampling block <b>370</b> and converted into an analog form by the DAC <b>372</b> and provided as the output signal <b>194</b> which in this case is an analog CVBS TV output signal.

0119The carrier recovery block <b>378</b> provides the analog mode frequency shift feedback signal <b>390</b> to adjust the amount of frequency rotation provided by the frequency rotator <b>350</b>. The adjustment is such that the frequency shift provided by the frequency rotator <b>364</b> aligns the picture carrier at DC. The carrier recovery block <b>378</b> is similar to a phase-locked loop and is discussed further in relation to <figref idref="DRAWINGS">FIG. 12A</figref>. However, in this exemplary embodiment, the carrier recovery block <b>378</b> also provides a phase control signal <b>404</b> to the phase rotators <b>376</b>, <b>382</b> and <b>386</b> to control the amount of phase rotation that is applied. A phase rotator is a complex mixer that adjusts the phase of its input to correct for phase errors. However, in some cases, these phase rotators can be optional. Accordingly, in alternative embodiments, the picture carrier recovery block <b>366</b> does not include phase rotators <b>376</b>, <b>382</b> and <b>386</b> and the carrier recovery block <b>378</b> does not provide the phase control signal <b>404</b>.

0120In this exemplary embodiment, the picture carrier recovery block <b>366</b> performs frequency correction to account for frequency offset errors in the processed digitized coarse channel signal <b>192</b> and phase noise compensation to compensate for phase perturbations to reduce phase noise in the processed digitized coarse channel signal <b>192</b>. The phase noise reduction includes compensating for phase perturbations which includes phase noise and systematic variations caused by spurs and the like. Typically, a television tuner imparts phase noise onto the incoming wideband television signal <b>10</b>. The phase noise typically appears as white noise in the processed television signal. If the phase noise is less than 200 kHz or so, the phase noise can be tracked and attenuated. Conventional television receivers do not track or compensate for phase noise. It should be understood that the phase noise reduction provided by the picture carrier recovery block <b>366</b> can be used with other television receivers that employ a different technique for filtering an input signal that must then be demodulated to obtain the desired television channel signal <b>22</b>. In these cases, the picture carrier recovery block <b>366</b> can be applied to the filtered input signal and the picture carrier recovery block <b>366</b> does not have to generate the analog mode frequency shift feedback signal <b>390</b>.

0121Accordingly, when operating in analog operation mode, there are actually two processing loops in the video processing block <b>182</b>. There is an outer loop including the frequency rotator <b>350</b>, video pre-polyphase filter <b>352</b><i>p</i>, video polyphase filter <b>352</b>, video filter <b>356</b>, digital VGA <b>358</b>, frequency rotator <b>364</b>, and the picture carrier recovery block <b>366</b> that acts like a frequency tracking loop. There is also an inner loop including the phase rotators <b>376</b>, <b>382</b> and <b>386</b>, and the carrier recovery block <b>378</b> that acts as a phase tracking loop. The frequency tracking loop tracks the carrier frequency of the desired television channel signal <b>22</b> and corrects for frequency offset errors to lock onto the picture carrier frequency of the desired television channel signal <b>22</b>. The phase tracking loop tracks and reduces the phase noise in the desired television channel signal <b>22</b>. The phase tracking loop requires a high bandwidth in order to react quickly to the phase noise. Accordingly, filters employed in the phase tracking loop have a reduced number of filter taps and other delays in this loop are kept at a minimum. The frequency lock loop cannot track the phase noise since it has too much delay due to the amount, and sharpness, of the filtering that is done.

0122The first frequency rotator <b>350</b>, video pre-polyphase filter <b>352</b><i>p</i>, first video polyphase filter <b>352</b>, video resampling control block <b>354</b>, video filter <b>356</b>, digital VGA <b>358</b>, and the second frequency rotator <b>364</b> can be considered to be elements of a signal processing pathway that cumulatively provide frequency shifting and filtering to remove extraneous signal components and output down-shifted frequency components of the desired television channel signal <b>22</b> component including the picture carrier signal frequency. Also the video pre-polyphase filter <b>352</b><i>p </i>and the first video polyphase filter <b>352</b> can be considered to be a video polyphase filter stage for resampling a signal at a new sampling rate. The picture carrier recovery block <b>266</b> then generally receives the down-shifted frequency components of the television channel signal component, locks onto the picture carrier signal frequency, and provides a demodulated television channel signal. In at least some embodiments, during the locking process, the operation of the picture carrier recovery block <b>366</b> can be modified when operating in an overmodulation handling mode to deal with the presence of overmodulation when locking onto the picture carrier.

0123The picture carrier recovery block <b>366</b> modifies its operation in the presence of overmodulation in the filtered picture carrier signal when tracking at least one of a frequency error signal and a phase error signal by applying a weight to at least one of the frequency error signal and the phase error signal or by using a previous correction value. The phase error signal is produced by comparing a phase of the filtered picture carrier signal with a phase reference signal. There can be embodiments in which only frequency tracking is employed by tracking the frequency error signal, only phase noise reduction is employed by tracking the phase error signal as well as embodiments in which both frequency and phase tracking is employed as shown in <figref idref="DRAWINGS">FIG. 12A</figref>. Accordingly, there can be embodiments in which the operation of the picture carrier recovery block <b>366</b> is modified in the presence of overmodulation while generating the analog mode frequency shift feedback signal <b>390</b> that is provided to the signal processing pathway. There can also be embodiments in which the operation of the picture carrier recovery block <b>366</b> is modified in the presence of overmodulation during the production of the phase control signal <b>404</b> to reduce phase noise. There can also be embodiments in which the operation of the picture carrier recovery block <b>366</b> can be modified during overmodulation while producing both the analog mode frequency shift feedback signal <b>390</b> and the phase control signal <b>404</b>. The description that follows is for the embodiment in which overmodulation handling is used for both the generation of the phase control signal <b>404</b> and the analog mode frequency shift feedback signal <b>390</b>, but it can be modified by applying overmodulation handling to the production of only one of these two signals as mentioned.

0124Referring now to <figref idref="DRAWINGS">FIG. 12A</figref>, shown therein is a block diagram of an exemplary embodiment of the carrier recovery block <b>378</b> which generally includes a phase correction stage, a frequency correction stage and a status stage. The phase correction stage processes the phase rotated output of the carrier recovery filter <b>374</b> and the magnitude level signal <b>402</b> to produce the phase control signal <b>404</b>. The frequency correction stage also processes the phase rotated output of the carrier recovery filter <b>374</b> and the magnitude level signal <b>402</b> to produce the analog mode frequency shift feedback signal <b>390</b>. The status stage receives phase and frequency error signals from the phase and frequency correction stages, respectively, to determine if there is phase lock and/or frequency lock. Each of these stages are now discussed in more detail.

0125The phase correction stage includes a cordic block <b>450</b>, a phase-frequency detector <b>452</b>, a lowpass filter <b>454</b>, a phase adjustment block <b>456</b>, a phase loop amplifier <b>458</b>, and a phase oscillator block <b>460</b>. The phase adjustment block <b>456</b> includes a phase inversion detector <b>462</b>, and a phase inversion block <b>464</b>. The phase oscillator block <b>460</b> includes a phase accumulator <b>466</b> and a cordic block <b>468</b>.

0126The cordic block <b>450</b> receives the I and Q signals of the phase rotated output of the carrier recovery filter <b>374</b> and produces a corresponding phase signal <b>470</b>. The phase-frequency detector <b>452</b> then processes the phase signal <b>470</b> to produce a phase error signal <b>472</b> by comparing the phase signal <b>470</b> with a phase reference signal expected for zero phase noise. The phase reference signal is typically a phase vector that lies along the I axis.

0127The phase-frequency detector <b>452</b> can operate in different phase tracking modes including a full phase tracking mode, and an overmodulation handling mode. The overmodulation handling mode can be a first overmodulation handling mode or a second overmodulation handling mode. The use of the overmodulation handling mode provides protection against overmodulation of the picture carrier signal for analog broadcast television signals. For instance, for NTSC television signals, negative modulation is used and an NTSC television signal has its highest signal level during the sync interval. However, in some cases, such as when very white television components are transmitted, due to overmodulation, the amplitude of the picture carrier signal can tend towards zero, be very small and less than the noise level, or can even undergo 180 degrees phase reversal. It then becomes difficult to track the phase and if one is not careful then noise can be tracked instead of the picture carrier. When the level of the real picture carrier then becomes larger and is detected, it can be out of phase with the phase that is currently being tracked. This overmodulation situation can be reflected in the magnitude level signal <b>402</b>, which can be used to obtain accurate results without phase synchronization.

0128<figref idref="DRAWINGS">FIG. 12B</figref> shows an exemplary signal and defines the percentage of video modulation. Most specifications define overmodulation as more than 87.5% of modulation as shown in <figref idref="DRAWINGS">FIG. 12B</figref>. However, some television signals are transmitted with an amount of modulation that is larger than 87.5% and in some cases can even have over 100% modulation. During portions of television signals in which the modulation is greater than 100%, the phase of the picture carrier is reversed.

0129The phase-frequency detector <b>452</b> can operate in the first overmodulation mode or the second overmodulation mode to compensate for overmodulation. Both of these modes are a non-coherent way to detect and handle overmodulation. Both of these modes employ a magnitude level, provided by the magnitude level signal <b>402</b>, which is the level of the filtered picture carrier signal to determine when overmodulation has occurred and to gate off unreliable phase information. Accordingly, the overmodulation filter <b>406</b> is a lowpass filter in which the cutoff frequency is set to remove rapid transitions and spurious switches in the picture carrier signal. The phase of the filtered picture carrier signal is not used in either overmodulation mode. Further, the input to the picture carrier recovery block <b>366</b> is used to detect overmodulation rather than the demodulated output (i.e. the output of the VSB filter <b>384</b>).

0130Referring to <figref idref="DRAWINGS">FIG. 12C</figref>, shown therein is a graphical representation of the first overmodulation handling mode. The value of the magnitude level signal <b>402</b> is compared against a threshold, Noise_thresh, to determine if overmodulation is occurring. When the magnitude level signal <b>402</b> falls below the threshold, the carrier recovery block <b>378</b> enters into a loop frozen mode and essentially holds the phase control signal <b>404</b> and the analog mode frequency shift feedback signal <b>390</b> constant by using the last or previous phase correction and frequency correction values prior to entering the loop frozen mode. This can also be referred to as free-running phase tracking. Accordingly, in this case, unreliable phase information measured by the phase-frequency detector <b>452</b> is prevented from being further processed by the phase and frequency correction stages when the magnitude of the filtered picture carrier signal is small and unreliable. When the magnitude level signal <b>402</b> is greater than the threshold, the frequency-phase detector <b>452</b> exits overmodulation handling mode, and the phase control signal <b>404</b> and the analog mode frequency shift feedback signal <b>390</b> become active again to actively track the phase and frequency errors. The full value of the phase and frequency errors are used and so it can be considered that a weight having a value of 1 is applied to these errors in this case.

0131The value of the threshold Noise_thresh can be selected to be the value that is expected for the white level of the magnitude level signal <b>402</b> when overmodulation is at about 90%; i.e. the magnitude level corresponding to a white level is 10% for 90% overmodulated video on a normalized basis, and the threshold Noise_thresh can be set to 0.1 in terms of normalized magnitude, i.e. the threshold Noise_thresh is at a magnitude level equivalent to a normalized magnitude level of about 10%. This normalization can be done by the overmodulation magnitude detector <b>408</b> or another suitable element. This threshold can also be changed depending on whether the phase and frequency correction stages (i.e. phase and frequency correction loops) are operating in an acquisition state and are attempting to determine coarse phase and frequency settings, or if they are operating in a lock state and have determined the current phase and frequency operating points. For instance, the threshold Noise_thresh can be set to a lower value during the acquisition state than during the lock state so that the carrier recovery tracks more often during the acquisition state. Alternatively, in other embodiments, the threshold Noise_thresh can have the same value during both acquisition and lock states.

0132In addition, the timing for applying the weights or using a previous phase or frequency correction value with respect to the detection of overmodulation can be skewed (i.e. advanced or retarded), or stretched. Accordingly, the carrier recovery block <b>378</b> can apply a guard band, which is a block of time, that can be skewed or stretched with respect to the onset and termination of overmodulation detection so that during the guard band the phase and frequency errors are not being actively tracked. The amount of skewing or stretching can be based on the severity of the noise at the output of the frequency rotator <b>364</b>. This will provide robustness under more severe phase noise conditions, such as under multipath scenarios. In some embodiments, the timing of the guard band for compensating for overmodulation earlier (in terms of the number of samples) with respect to the onset of overmodulation and to continue compensating for overmodulation for some time afterwards (once again in terms of the number of samples) with respect to the termination of overmodulation can be the same. The number of samples to advance or delay entry into or exit out of overmodulation compensation with respect to the onset or termination of overmodulation detection can be in the range of 8 to 15 samples. The guard band can also be used in the second overmodulation handling mode which is now discussed.

0133Referring to <figref idref="DRAWINGS">FIG. 12D</figref>, shown therein is a graphical representation of the second overmodulation handling mode. The second overmodulation handling mode uses a soft approach to gate off unreliable phase information which would otherwise be measured by the phase-frequency detector <b>452</b> during overmodulation. The second overmodulation handling mode applies a weight to the phase error signal <b>472</b> and the frequency error signal <b>494</b> before these signals are used in the remainder of the phase and frequency correction stages.

0134In some embodiments, the weighting is a piece-wise linear function based on the value of the magnitude level signal <b>402</b> as shown in <figref idref="DRAWINGS">FIG. 12D</figref>. The maximum weight is 1, which is similar to the full phase tracking mode when the measured phase and frequency errors are used to actively track phase and frequency in the frequency and phase correction stages. Below the threshold Noise_thresh, operation is somewhat similar to the loop frozen operation for the first overmodulation handling mode in which the phase control signal <b>404</b> and the analog mode frequency shift feedback signal <b>390</b> are held constant by applying a weight of zero to the phase error signal <b>472</b> and the frequency error signal <b>494</b>. In this case, the output of the lowpass filter <b>454</b> decays to zero depending on the duration of the loop frozen operation. Generally, a large weight is applied to the phase error signal <b>472</b> and the frequency error signal <b>494</b> when the magnitude level signal <b>402</b> is large, and a small weight is applied to the phase error signal <b>472</b> and the frequency error signal <b>494</b> when the magnitude level signal <b>402</b> is small. Operation in the sloped regions can be referred to as soft phase tracking.

0135The shape of the piece-wise weighting function can be changed by setting different values for the thresholds Noise_thresh, OM<b>1</b>_thresh, and OM<b>2</b>_thresh, the first weight, the second weight, and by using different values for the two slopes. In alternative embodiments, more than three threshold values can be used, and the additional thresholds are used to determine the amount of soft phase tracking in finer granularities. Also, different values can be selected for these parameters if the phase and frequency correction stages (i.e. phase and frequency correction loops) are operating in an acquisition state and are attempting to determine coarse phase and frequency settings, as compared to when these stages are operating in a lock state and have determined the current phase and frequency operating points.

0136In this exemplary embodiment, the slope of the weighting curve increases with magnitude until the maximum weighting is reached. The first weight is set at noise_w, and the threshold Noise_thresh is in the range of a magnitude level that is equivalent to a normalized magnitude level of about 5% to 10%. The threshold OM<b>1</b>_thresh is chosen to be greater than the threshold Noise_thresh and is generally less than a magnitude level that is equivalent to a normalized magnitude level of 0.3. The second weight is chosen to be greater than the first weight and generally less than a magnitude level that is equivalent to a normalized magnitude level of 0.2. The threshold OM<b>2</b>_thresh is chosen to be greater than the second threshold and is less than a magnitude level that is equivalent to a normalized magnitude level of 1.0. The slopes are chosen so that the weighting curve is a piece-wise linear function that connects the different threshold/weighting pairs together. In general, the weighting function can be the same during the acquisition and lock states. However, for more phase tracking during the acquisition state, the thresholds can be set lower and the weights can be set higher than the settings for the thresholds and weights used in the lock state. Timing changes, similar to those discussed in the first overmodulation handling mode may or may not be used in the second overmodulation handling mode.

0137In all of the phase tracking modes, the phase correction loop operates with the gain value G_phbeing selected for the phase loop amplifier <b>458</b> (which can be implemented as a multiplier) to provide a compromise between the bandwidth of the phase tracking stage and hence the speed with which the phase tracking stage can lock onto the phase reference signal and respond to changes in the phase error signal <b>472</b>. The amount of filtering provided by the lowpass filter <b>454</b> can also be adjusted to obtain a desired speed and bandwidth for the phase tracking stage. Also, the gain value G_phis selected to provide an acceptable amount of phase error while the phase tracking stage still remains in lock. The value of G_phcan also be modified during operation depending on whether phase or frequency lock has occurred to improve the operation of the phase tracking stage. As a rule of thumb, the gain and loop bandwidth are selected so that under a step response condition, the residual phase error does not result in multiple oscillations before settling to zero.

0138In the linear regions of the second overmodulation handling mode, the phase-frequency detector <b>452</b> generally operates by multiplying the phase error with a positive non-zero weight that is less than 1, so that the effective gain value G_phof the phase correction loop is set to a smaller value than that used in the full phase tracking mode. This allows the phase tracking stage to react more slowly to changes in the phase error signal <b>472</b> since the phase error signal <b>472</b> is more likely to be affected by noise due to the lower value of the magnitude level signal <b>402</b>.

0139In each of the phase tracking modes, the phase error signal <b>472</b> is filtered by the lowpass filter <b>454</b> to obtain a filtered phase error signal <b>474</b>. The filtered phase error signal <b>474</b> is amplified by the loop gain and passed into the phase adjustment block <b>456</b>. The filtered phase error signal <b>474</b> is processed by the phase adjustment block <b>456</b> which compensates for 180 degrees phase synchronization that may occur in the filtered phase error signal <b>474</b>. The 180 degrees phase synchronization may occur during system initialization under very high signal to noise conditions, where the picture carrier recovery block <b>366</b> can lock onto the 180 degrees operating point. In this exemplary embodiment, the phase adjustment block <b>456</b> can also be disabled or enabled.

0140The phase inversion detector <b>462</b> checks the filtered phase error signal <b>474</b> to determine if there is a +/−180 degrees phase shift in the phase error. The phase inversion detector <b>462</b> checks for how long the filtered phase error signal <b>474</b> is in the region of +/−180 degrees (this can be done using a counter). If the filtered phase error signal <b>474</b> is in the region of +/−180 degrees for a certain period of time, the phase loop is running at an operating point that is off by 180 degrees. The phase inversion detector <b>462</b> then produces a 180 degrees phase inversion detection signal <b>476</b> to indicate whether this 180 degrees phase synchronization has occurred. The filtered phase error signal <b>474</b> is also amplified by the phase loop amplifier <b>458</b> to produce an amplified phase error signal <b>478</b>. The phase inversion block <b>464</b> receives both the 180 degrees phase inversion detection signal <b>476</b> and the amplified phase error signal <b>478</b>. If 180 degrees phase synchronization has not been detected, then the output of the phase adjustment block <b>456</b> is the amplified phase error signal <b>478</b>; i.e. no correction is applied by the phase inversion block <b>464</b>. If 180 degrees phase synchronization has been detected, then the output of the phase adjustment block <b>456</b> is a 180 degrees phase adjusted signal; i.e. a 180 degree phase correction is provided by the phase inversion block <b>464</b>.

0141The phase oscillator block <b>460</b> produces the phase control signal <b>404</b> based on the output of the phase adjustment block <b>456</b>. The phase accumulator <b>466</b> accumulates the values of the amplified phase error signal <b>478</b>, which has been corrected for 180 degrees phase synchronization, to obtain an integrated phase value. This accumulation, or integration, which can be weighted in some implementations, allows the phase tracking stage to respond a bit slower to instantaneous changes in the phase error signal <b>472</b> and therefore operate in a more stable fashion. The integrated phase value is then converted to I and Q signals by the cordic block <b>468</b> which are outputted as the phase control signal <b>404</b> and provided to the phase rotators <b>376</b>, <b>382</b> and <b>386</b>. When the integrated phase error tends towards zero, the phase tracking stage is locked and the phase noise in the desired television channel signal <b>22</b> is compensated.

0142The frequency correction stage also includes the cordic block <b>450</b>, the phase-frequency detector <b>452</b>, a summer <b>480</b>, a decimation filtering block <b>482</b>, a frequency loop amplifier <b>484</b>, and a frequency oscillator block <b>486</b>. The frequency oscillator block <b>486</b> includes a frequency accumulator <b>488</b>, a frequency clipping block <b>490</b>, a phase accumulator <b>510</b>, and a cordic block <b>492</b>. The frequency correction stage can also operate in a full frequency tracking mode, and employ soft frequency tracking and free running frequency tracking, as in the case of the phase correction stage. The frequency correction stage is a slower loop than the phase correction stage. Accordingly, the value for the frequency loop gain −G_frand the bandwidth of the frequency loop is selected so that the step response for this loop is dampened with little overshoot. The frequency correction stage should be able to track within 500 kHz of the actual picture carrier frequency taking into account the frequency offset error.

0143The phase-frequency detector <b>452</b> generates a frequency error signal <b>494</b>. The frequency error signal <b>494</b> is derived from the phase error signal <b>472</b> in that the values in the frequency error signal <b>494</b> are a series of delta-phase errors since frequency is the derivative of phase. The frequency error signal <b>494</b> indicates the offset of the picture carrier frequency from DC at the output of the frequency rotator <b>364</b>. The phase rotators <b>376</b>, <b>382</b> and <b>386</b> can correct for some amount of frequency error before the signal enters into the carrier recovery block <b>378</b>. As a result, the amount of frequency correction that is required by the frequency loop takes into account the amount of frequency error to be corrected by the phase rotators <b>376</b>, <b>382</b> and <b>386</b>. This information is considered by adding the frequency error signal <b>494</b> to the output of the phase adjustment block <b>456</b> via the summer <b>480</b> to produce an adjusted frequency error signal <b>496</b>.

0144The adjusted frequency error signal <b>496</b> is then filtered and decimated by the decimation filtering block <b>482</b> to produce a filtered frequency error signal <b>498</b>. The decimation filtering block <b>482</b> provides lowpass filtering to smooth out the values in the adjusted frequency error signal <b>496</b>. Decimation, which is optional, is used in this exemplary embodiment for increasing implementation efficiency. The filtered frequency error signal <b>498</b> is then amplified by the frequency loop amplifier <b>484</b>, which can also be implemented as a multiplier, to produce an amplified frequency error signal <b>500</b>. The amount of filtering provided by the decimation filtering block <b>482</b> and the amount of amplification provided by the frequency loop amplifier <b>484</b> can be adjusted to control the bandwidth and hence the speed of the frequency tracking stage.

0145The amplified frequency error signal <b>500</b> is then provided to the frequency oscillator block <b>486</b>, which generates the analog mode frequency shift feedback signal <b>390</b> based on the amplified frequency error signal <b>500</b>. The amplified frequency error signal <b>500</b> is first processed by the frequency accumulator <b>488</b>, which keeps track of the current frequency and updates it with the current value in the amplified frequency error signal <b>500</b> to produce a frequency adjusted signal <b>502</b>. In some implementations, the frequency accumulator <b>488</b> can average consecutive values in the amplified frequency error signal <b>500</b> prior to adjusting the current frequency value. In some cases, weighted averaging can be used. The amplified frequency error signal <b>500</b> allows for compensating for the frequency offset error that was discussed previously. The frequency accumulator <b>488</b> is provided with an initial frequency value, which is the amount of frequency shift that is expected to be applied to the processed digitized coarse channel signal <b>192</b> to center it about DC. At initial operation, the current frequency value is set based on the initial frequency value and thereafter updated based on the values in the amplified frequency error signal <b>500</b>.

0146The frequency clipping block <b>490</b> specifies upper and lower limits for the frequency adjusted signal <b>502</b> to define a range of frequencies over which frequency tracking operates for picture carrier recovery. The amplified frequency error signal <b>500</b> is added to the current frequency in the frequency accumulator <b>488</b>, and the result is compared to a maximum and minimum frequency in the frequency clipping block <b>490</b>. If the resulting frequency is greater than the maximum or less than the minimum clipper frequencies, the frequency accumulator value is clipped (i.e. limited) to the maximum value limit or the minimum value limit, respectively; hence the feedback signal <b>512</b> from the frequency clipping block <b>490</b> to the frequency accumulator <b>488</b>. In alternative embodiments, the clipping function can be replaced by a wrapping function in which, when one frequency limit is reached without the picture carrier being locked, the current frequency used by the frequency accumulator <b>488</b> is set to the opposite limit via the feedback connection. The output of the frequency clipping block <b>490</b> is provided to the phase accumulator <b>510</b> which in turn provides an input to the cordic block <b>492</b> which then generates the analog mode frequency shift feedback signal <b>390</b>. The phase accumulator <b>510</b> keeps track of the current phase. The frequency accumulator <b>488</b> provides the value by which the phase accumulator <b>510</b> is incremented on each cycle after processing by the frequency clipping block <b>490</b>. The higher the frequency, the more rapidly the phase will accumulate.

0147The status stage includes a lock detector <b>504</b>. The lock detector <b>504</b> receives the filtered phase error signal <b>474</b> and the filtered frequency error signal <b>498</b> and determines whether lock has occurred for phase and frequency tracking. The lock detector <b>504</b> provides a phase lock status signal <b>506</b> and a frequency lock status signal <b>508</b>. These values can be stored in status registers associated with the video processing block <b>182</b>. The values can then be used to modify some parameters in the phase and frequency tracking stages, as well as some parameters of the blocks in the picture carrier recovery block <b>366</b> such as the bandwidth of the carrier recovery filter <b>374</b> and the AGC filter <b>380</b>. The video processing block <b>182</b> can adopt a different set of parameters during phase or frequency acquisition versus phase or frequency lock (i.e. for implementing coarser or tighter search ranges as well as faster or slower response). In at least some embodiments, the lock detector <b>504</b> can also generate signal B<b>1</b>, which is used to indicate that a lock has been made to the picture carrier, and communicate signal B<b>1</b> to the control block <b>190</b> as explained previously with respect to <figref idref="DRAWINGS">FIG. 7</figref>.

0148As mentioned previously, various audio standards are used for the audio information that is present in analog television signals. For instance in North America, there is only one audio carrier that is used with a television signal, however, the audio carrier may carry stereo audio information. In Europe, a NICAM standard can be used which is a digitally encoded audio signal that is included with an analog television signal. However, other standards use two analog audio carrier signals to encode “right-sided” audio information and “left-sided” audio information. Each of these scenarios can be handled by the first and second audio filtering blocks <b>184</b> and <b>186</b>, and the audio processing block <b>188</b>. If only one audio carrier signal is used, then just the first audio filtering block <b>184</b> is enabled. Digital broadcast standard television signals include multiplexed audio and video information. In this exemplary embodiment, such television signals are processed by the video processing block <b>182</b>, which provides a digital output containing the video and audio information as the output signal. The video and audio information can then be further processed by another element, such as a downstream digital demodulator (not shown), for example. The processing provided by the first and second audio filtering blocks <b>184</b> and <b>186</b>, and the audio processing block <b>188</b> can provide SIF (a Sound Intermediate Frequency signal) or baseband sound outputs. With an SIF signal output, another audio decoder can be connected to the receiver <b>100</b> to process the audio information.

0149The structure of the first and second audio filtering blocks <b>184</b> and <b>186</b> are similar. Accordingly, only the first audio filtering block <b>184</b> will be discussed in greater detail with reference to a first embodiment shown in <figref idref="DRAWINGS">FIG. 13A</figref> and an alternative embodiment shown in <figref idref="DRAWINGS">FIG. 13B</figref>. The sound carrier signal is a narrowband signal that is separated from the accompanying video information (refer to <figref idref="DRAWINGS">FIG. 11B</figref> for an example). The first audio filtering block <b>184</b> is provided with the same processed digitized coarse channel signal <b>192</b> as the video processing block <b>182</b> or with another signal as explained below in relation to <figref idref="DRAWINGS">FIG. 13B</figref>. Audio signals can have a wide variation in bandwidth; the audio bandwidth range extends from about 50 kHz to 700 kHz. Accordingly, the first audio filtering block <b>184</b> employs a somewhat similar processing methodology as the video processing block <b>182</b> in that a fixed filter is used along with resampling to make it appear as if the fixed filter has a variable bandwidth. In addition, the first audio filtering block <b>184</b> generally employs a second frequency tracking loop that is configured to extract the audio carrier frequency of the desired television channel signal <b>22</b> for analog television broadcast standards and the first audio filtering block <b>184</b> is configured to apply a second known frequency shift to compensate for a known frequency offset in the audio carrier frequency. However, in alternative embodiments, the frequency tracking in the first audio filtering block <b>184</b> can be slaved to the frequency tracking employed by the video processing block <b>182</b> as described with relation to <figref idref="DRAWINGS">FIG. 13B</figref>.

0150Referring now to <figref idref="DRAWINGS">FIG. 13A</figref>, shown therein is an exemplary embodiment of the first audio filtering block <b>184</b>. The first audio filtering block <b>184</b> includes a frequency rotator <b>550</b>, a first decimation filtering block <b>552</b>, an audio pre-polyphase filter <b>554</b><i>p</i>, a first audio polyphase filter <b>554</b>, an audio resampling phase control block <b>556</b>, a second decimation filtering block <b>558</b>, a third decimation filtering block <b>560</b>, a multiplexer <b>562</b>, an audio filter <b>564</b>, an audio polyphase filter <b>566</b>, a frequency demodulator <b>568</b> and an audio IF carrier recovery block <b>570</b>. The audio pre-polyphase filter <b>554</b><i>p </i>and the first audio polyphase filter <b>554</b> can be considered to be an audio polyphase filter stage for resampling a signal at a new sampling rate. The decimation filtering block <b>552</b>, audio pre-polyphase filter <b>554</b><i>p</i>, first audio polyphase filter <b>554</b>, audio resampling phase control block <b>556</b>, second decimation filtering block <b>558</b>, third decimation filtering block <b>560</b>, multiplexer <b>562</b>, audio filter <b>564</b>, and audio polyphase filter <b>566</b> can be referred to as an audio filter stage. Also, the second frequency tracking loop includes the frequency rotator <b>550</b>, the audio filter stage, the frequency demodulator <b>568</b> and the audio IF carrier recovery block <b>570</b>.

0151The frequency rotator <b>550</b> receives the processed digitized coarse channel signal <b>192</b> and shifts the frequency content of this signal such that the frequency content of the audio of the desired television channel signal is approximately centered about DC. However, due to the frequency offset uncertainty, exact centering about DC is likely not achieved. The output of the frequency rotator <b>550</b> is then filtered and downsampled by the decimation filtering block <b>552</b>. The output of the decimation filtering block <b>552</b> is filtered by the audio pre-polyphase filter <b>554</b><i>p</i>, and subsequently resampled by the audio polyphase filter <b>554</b> based on a first resampling control signal <b>572</b>. The audio pre-polyphase filter <b>554</b><i>p </i>is configured and used in a similar manner as the video pre-polyphase filter <b>352</b><i>p</i>. It should be noted that the function of the decimation filtering block <b>552</b> can optionally be included into the functionality of the audio pre-polyphase filter <b>554</b><i>p </i>and/or the audio polyphase filter <b>554</b>. The audio resampling phase control block <b>556</b> provides the value for the first resampling control signal <b>572</b> based on the audio broadcast standard that corresponds with the television broadcast transmission standard for the desired television channel signal. The audio polyphase filter <b>554</b> then resamples the output of the audio pre-polyphase filter <b>554</b><i>p </i>so that its bandwidth is transformed to match the bandwidth of the desired audio signal with the bandwidth of the fixed audio filter <b>564</b>; this operation is analogous to that in the video processing block <b>182</b> and therefore does not need to be further discussed. In alternative embodiments, the functionality of the audio resampling phase control block <b>556</b> can be provided by the control block <b>190</b>.

0152However, since the bandwidth of the desired audio signal ranges from 50 kHz to 700 kHz, and a relatively large sampling rate is being used, the first audio filtering block <b>184</b> employs the decimation filtering blocks <b>558</b> and <b>560</b> and the multiplexer <b>562</b> for more efficient processing. These blocks are used to extend the range of bandwidth control by an additional 2 octaves. The blocks <b>558</b> to <b>562</b> are not required if the audio polyphase filter <b>554</b> is configured to deal with these different frequency ranges for the audio.

0153In this exemplary embodiment, there are three audio signal pathways from the audio polyphase filter <b>554</b> to the audio filter <b>564</b> via the multiplexer <b>562</b>. The audio resampling phase control block <b>556</b> provides an audio pathway selection control signal <b>574</b> to the multiplexer <b>562</b> to select one of the three audio pathways. A first audio pathway exists from the output of the audio polyphase filter <b>554</b> to the audio filter <b>564</b>. A second audio pathway exists from the output of the audio polyphase filter <b>554</b> through the decimation filtering block <b>558</b> to the audio filter <b>564</b>. A third audio pathway exists from the output of the audio polyphase filter <b>554</b> through the decimation filtering blocks <b>558</b> and <b>560</b> to the audio filter <b>564</b>.

0154The first audio pathway does not provide any downsampling, while the second audio pathway provides a first amount of downsampling and the third audio pathway provides a second amount of downsampling that is larger than the first amount of downsampling. Accordingly, the first audio pathway can be selected when the desired audio signal has a high bandwidth at the upper end of the audio bandwidth range. The second audio pathway can be selected when the desired audio signal has a medium bandwidth that is somewhere between the lower and upper limits of the audio bandwidth range. The third audio pathway can be selected when the desired audio signal has a small bandwidth that is at the lower end of the audio bandwidth range. In general, a greater or lesser number of decimation stages may be configured depending on the desired range of audio bandwidths to be supported.

0155The audio filter <b>564</b> operates in a similar manner as the video filter <b>356</b> and therefore does not need to be described in detail. The output of the audio filter <b>564</b> is provided to the audio polyphase filter <b>566</b> which upsamples the audio signal to generate a sound IF (SIF) signal <b>576</b>. The SIF signal <b>576</b> can be further processed by the audio processing block <b>188</b>. The output of the audio polyphase filter <b>566</b> can also be sent to the frequency demodulator <b>568</b> which demodulates this output to produce the intermediate audio signal <b>196</b>. The frequency demodulator <b>568</b> is typically an FM demodulator which is well known to those skilled in the art. If the desired audio signal is a mono audio signal, then the intermediate audio signal <b>196</b> is a baseband audio signal. For other audio broadcast standards, the intermediate audio signal <b>196</b> is another modulated audio signal. The sampling rate associated with the SIF <b>576</b> signal and the intermediate audio signal <b>196</b> can be on the order of 1.536 MHz to improve noise performance.

0156The audio IF carrier recovery block <b>570</b> receives a sound IF carrier recovery signal <b>578</b> from the audio processing block <b>188</b> (discussed with relation to <figref idref="DRAWINGS">FIG. 14</figref>), which is used to track the audio carrier signal. In alternative embodiments, the audio IF carrier recovery block <b>570</b> can receive the output signal of the frequency demodulator <b>568</b>. However, the sound IF carrier recovery signal <b>578</b> is a better quality signal with less noise. The audio IF carrier recovery block <b>570</b> tracks an audio carrier signal that corresponds to the desired television channel signal <b>22</b> and provides an audio frequency shift feedback signal <b>580</b> to the frequency rotator <b>550</b> for shifting the audio carrier frequency to DC when doing baseband demodulation and for shifting frequency content of the audio information to DC for SIF only processing. In an alternative embodiment, the audio IF carrier recovery block <b>570</b> is not dependent on the sound IF carrier recovery signal <b>578</b> but rather is configured to operate in a free running mode such that frequency rotator <b>550</b> provides a fixed frequency shift. In this case, audio carrier recovery may be performed at a later stage without feedback to frequency rotator <b>550</b> or to audio IF carrier recovery block <b>570</b>. The implementation of the audio IF carrier recovery block <b>570</b> is known to those skilled in the art.

0157In an alternative embodiment, the audio filtering blocks <b>184</b> and <b>186</b> have a different configuration as shown by audio filtering block <b>184</b>′ in <figref idref="DRAWINGS">FIG. 13B</figref>. The input to the audio filtering block <b>184</b>′ is the output of the frequency rotator <b>350</b> in the video processing block <b>182</b> instead of the processed digitized coarse channel signal <b>192</b>. The operation of the audio IF carrier recovery block <b>570</b> is also modified to compensate for the frequency shift provided in the output of the frequency rotator <b>350</b>. Furthermore, in order that the audio signal may benefit from the phase error tracking provided by the carrier recovery block <b>378</b> of the video processing block <b>182</b>, the output of the phase accumulator <b>510</b> is used to produce the audio frequency shift signal <b>580</b>′ which controls the frequency shift applied by the frequency rotator <b>550</b> to its input signal: the output of the frequency rotator <b>350</b>. This can be achieved by adding the output of the phase accumulator <b>510</b> in the carrier recovery block <b>378</b> to the output of a similar phase accumulator (not shown) in the audio IF carrier recovery block <b>570</b>. Accordingly, the audio IF carrier recovery block <b>570</b> still generates the free running frequency that is applied to the frequency rotator <b>550</b> in terms of frequency, but with phase correction provided by the output of the phase accumulator <b>510</b> to generate the audio frequency shift signal <b>580</b>′. The end result is that the output of the frequency rotator <b>550</b> is shifted to DC in the same way that it would have been if the input to the frequency rotator <b>550</b> had not come from the output of the frequency rotator <b>350</b> (i.e. as currently described in <figref idref="DRAWINGS">FIG. 13A</figref>). In this way, audio carrier recovery is effectively slaved to picture carrier recovery. Note that considerable processing delay may exist in the video path between the output of the frequency rotator <b>350</b> and the output of the phase accumulator <b>510</b>. Additional benefit to the reduction of phase noise in the audio path may be achieved if a similar amount of delay is inserted between the output of the frequency rotator <b>350</b> and the input to the frequency rotator <b>550</b>. In this way the frequency correction provided by the frequency rotator <b>350</b> and the phase correction provided by the phase accumulator <b>510</b> are effectively synchronized.

0158Referring now to <figref idref="DRAWINGS">FIG. 14</figref>, shown therein is a block diagram of an exemplary embodiment of the audio processing block <b>188</b>. The audio processing block <b>188</b> has a first processing pathway including a first decimation filtering block <b>600</b>, a de-emphasis filter <b>602</b>, a first multiplexer <b>604</b>, and a first audio polyphase filter <b>606</b>. The audio processing block <b>188</b> also includes a second processing pathway including a second decimation filtering block <b>608</b>, a de-emphasis filter <b>610</b>, a second multiplexer <b>612</b> and a second audio polyphase filter <b>614</b>.

0159The first and second decimation filtering blocks <b>600</b> and <b>608</b> each have first and second stages that perform decimation filtering. The output of the first stage of the first decimation filtering block <b>600</b> is connected to a pilot recovery and audio extraction block <b>616</b>. The output of the pilot recovery and audio extraction block <b>616</b> is connected to the input of the second stage of the second decimation filtering block <b>608</b>.

0160The outputs of the first and second audio polyphase filters <b>606</b> and <b>614</b> are connected to a mixture block <b>618</b>. The outputs of the first and second audio polyphase filters <b>606</b> and <b>614</b> are also connected to frequency rotators <b>620</b> and <b>622</b>, respectively, which are both connected to a summer <b>624</b>. The output of the mixture block <b>618</b> and the summer block <b>624</b> are provided to a third multiplexer <b>626</b>. The third multiplexer <b>626</b> is connected to a third audio polyphase filter <b>628</b>. The audio processing block <b>188</b> also includes a NICAM processing block <b>630</b>, a FIFO <b>632</b> and a NICAM sampling control block <b>634</b>. The NICAM processing block <b>630</b> is a combination of a NICAM demodulator and decoder.

0161In operation, the audio processing block <b>188</b> can be provided with a variety of input signals depending on the audio broadcast standard that is used for providing the audio information of the desired television channel signal <b>22</b>. For instance, if one audio carrier is used for the desired television channel signal <b>22</b>, the first decimation filtering block <b>600</b> is provided with the first audio intermediate audio signal <b>196</b>. Alternatively, if two audio carriers are used for the desired television channel signal <b>22</b>, the first and second decimation filtering blocks <b>600</b> and <b>608</b> are provided with the first and second intermediate audio signals <b>196</b> and <b>198</b> respectively. The first and second de-emphasis filters <b>602</b> and <b>610</b> can also be provided with NICAM data which is further described below.

0162The first stage of each decimation filtering block <b>600</b> and <b>608</b> provides a first amount of filtering and downsampling to a first sampling rate such that secondary audio program (SAP) and L-R (Left-Right) audio information is retained. The SAP and L-R audio information exists from about 15 KHz up to about 90 KHz. The second stage of each decimation filtering block <b>600</b> and <b>608</b> provides a second amount of filtering and downsampling to a second sampling rate such that the SAP and L-R audio information is removed. The second sampling rate is a quarter of the first sampling rate. The output of the second stages of the decimation filtering blocks <b>600</b> and <b>608</b> is an FM demodulated audio baseband signal. The output of the second stage of the first decimation filtering stage <b>600</b> provides the sound IF carrier recovery signal <b>578</b> to the first audio filtering block <b>184</b>. Likewise, the output of the second stage of the second decimation filtering stage <b>608</b> provides a sound IF carrier shift signal <b>578</b>′ to the second audio filtering block <b>186</b>.

0163Any DC level in the sound IF carrier recovery signal <b>578</b>, which is an FM demodulated audio baseband signal, indicates that the mixing frequency of the frequency rotator <b>550</b> isn't exactly set to the carrier frequency. This can be understood since FM modulation changes the carrier frequency based on the instantaneous level of the modulating signal. If the modulating signal (at the transmitter) was at a DC level, this would be indistinguishable from a non-modulated carrier signal at a slightly different frequency. Accordingly, any DC that is present in the sound IF carrier recovery signal <b>578</b> must be due to the mixing frequency of the frequency rotator <b>550</b> being different from the actual transmitted carrier frequency and the audio carrier IF recovery block <b>570</b> accounts for this difference in the audio frequency shift feedback signal <b>580</b>.

0164The pilot recovery and audio extraction block <b>616</b> receives the SAP and L-R audio information from the first stage of the decimation filtering block <b>600</b> and locks to the pilot tone to demodulate the L-R audio information. The pilot recovery and audio extraction block <b>616</b> can also provide a locked carrier for SAP demodulation. The demodulated L-R audio information (i.e. BTSC L-R audio signal <b>638</b>) enters the second stage of the decimation filtering block <b>608</b>, and gets filtered and downsampled to 48 KHz, with content up to only about 15 KHz. Those skilled in the art are familiar with the implementation of the pilot recovery and audio extraction block <b>616</b>.

0165SAP and L-R audio information are present in North American BTSC signals (this audio standard somewhat corresponds to the NTSC video standard). Broadcasters in North America don't have to transmit SAP or stereo information. With other standards in other countries, the FM signal may contain other information, or have that information in another format. For example, in Japan, the L-R audio information sits in the FM demodulated signal just above the L+R audio information, similar to the BTSC signal, but is FM modulated instead of AM modulated. Some standards also transmit a “mode tone” within the FM demodulated signal, just above the audio portion. This can be used to indicate if a stereo signal is present, or that perhaps a second language is being transmitted.

0166FM audio signals are broadcast with pre-emphasis which boosts the high frequency content of the FM audio signals above high frequency noise which is inherent in FM transmission. Accordingly, the de-emphasis filters <b>602</b> and <b>610</b> are used to perform the opposite operation, de-emphasis or an attenuation of high frequencies, to restore the frequency content of the FM audio signal to its original levels. The L-R audio information is broadcast using a more complicated pre-emphasis function, requiring a corresponding de-emphasis function (wDBX) that removes the pre-emphasis. The de-emphasis functions performed by the de-emphasis filters <b>602</b> and <b>610</b> are known to those skilled in the art.

0167The output of the pilot recovery and audio extraction block <b>616</b> can provide SAP and EIAJ (the Japanese standard) L-R audio information. The EIAJ audio signal is FM modulated within the FM demodulated spectrum, above the baseband audio information. The second audio filtering block <b>186</b> can be used to FM demodulate this signal. For example, in EIAJ, the first audio filtering block <b>184</b> demodulates the broadcast FM signal which contains audio information from 0-15 KHz, FM modulated audio information centered at ˜31 KHz, and additional audio information. This signal gets routed to the second audio filtering block <b>186</b> to be FM-demodulated in order to extract the additional audio information.

0168The inputs of the multiplexer <b>604</b> are the output of the de-emphasis filter <b>602</b> and the SIF<b>1</b> signal provided by the first audio filtering block <b>184</b>. Likewise, the inputs of the multiplexer <b>612</b> are the output of the de-emphasis filter <b>610</b> and the SIF<b>2</b> signal provided by the second audio filtering block <b>186</b>. Either the SIF<b>1</b> and SIF<b>2</b> signals are selected as the outputs of the multiplexers <b>604</b> and <b>612</b> or the outputs of the de-emphasis filters <b>602</b> and <b>610</b> are selected according to a selection control input <b>640</b> that can be pre-defined or user defined.

0169The outputs of the multiplexers <b>604</b> and <b>612</b> are then provided to the audio polyphase filters <b>606</b> and <b>614</b> which restore the sampling rate to a power of 2 division of the clock rate of the ADC <b>106</b>. However, when NICAM audio signals accompany the desired television channel signal <b>22</b>, the re-sampling rate of the audio polyphase filters <b>606</b> and <b>614</b> is selected in a different manner as is described in further detail below.

0170The outputs of the audio polyphase filters <b>606</b> and <b>614</b> are provided to the mixture block <b>618</b>. The mixture block <b>618</b> merges the 2 audio channel signals appropriately to generate a 2 channel output (i.e. a left and right output).

0171Alternatively, when the audio processing block <b>188</b> provides an SIF audio output, the SIF<b>1</b> and SIF<b>2</b> audio signals are selected by the multiplexers <b>604</b> and <b>612</b> respectively, resampled by the audio polyphase filters <b>606</b> and <b>614</b> respectively and provided to frequency rotators <b>620</b> and <b>622</b>. The SIF<b>1</b> and SIF<b>2</b> audio signals are then shifted in frequency according to the IF frequency shift signals <b>642</b> and <b>644</b>, and are then summed by the summer <b>624</b>. Both frequency rotators <b>620</b> and <b>622</b> are enabled when two sound carriers are broadcast with the desired television channel signal <b>22</b>.

0172The multiplexer <b>626</b> selects between the output of the mixture block <b>618</b> and the output of the summer block <b>624</b> based on a selection control signal <b>646</b> which can be pre-defined or user-programmed depending on the mode of operation of the universal television receiver <b>100</b> which dictates the type of audio information that it should be providing. Alternatively, in at least some cases, the control signals can be provided by the control block <b>190</b> when the broadcast transmission standard has been detected. The multiplexer <b>626</b> selects the output of the mixture block <b>618</b> when the audio processing block <b>188</b> is configured to output an audio baseband signal. The multiplexer <b>626</b> selects the output of the summer <b>624</b> when the audio processing block <b>188</b> is configured to output an SIF audio signal. In both cases, the output of the multiplexer <b>626</b> is provided to the audio polyphase filter <b>628</b>, which resamples the audio information to correspond with the clock rate of the ADC <b>106</b> and produces the output audio signal <b>200</b>. Accordingly, the output of the audio polyphase filter <b>628</b> is either a baseband audio signal, which can be mono or stereo determined by the mixture block <b>618</b>, or an SIF signal which can consist of up to two FM carriers, at some programmable frequencies.

0173NICAM is a digital audio transmission standard. The NICAM processing block <b>630</b> processes one of the SIF<b>1</b> and SIF<b>2</b> signals and locks to the symbol rate and extracts and decodes the transmitted digital data to produce the decoded NICAM audio signal <b>636</b>. Those skilled in the art are familiar with NICAM demodulation and decoding. However, the symbol timing/period for the symbol rate is defined at the transmitter, which sent the desired television channel signal, and is unrelated to the sampling rates used in the receiver <b>100</b>. Nevertheless, the decoded NICAM audio signal <b>636</b> must be correctly output by the DAC block <b>110</b>, which is asynchronous to the transmitter, at the same rate.

0174The correct output rate can be determined by the NICAM sampling control block <b>634</b> via the FIFO <b>632</b>. The output of the FIFO <b>632</b> is then provided to the de-emphasis filters <b>602</b> and <b>610</b>. The FIFO <b>632</b> is a data structure and in alternative embodiments can be replaced with a memory element such as the on-chip memory (not shown) of the digital processing block <b>108</b>. The correct output rate can be determined by observing the “fullness” of the FIFO <b>632</b>. For instance, if the FIFO <b>632</b> is more than half-full with decoded NICAM audio data, then the output rate of the decoded NICAM audio data is too slow. In this case, the NICAM sampling control block <b>634</b> can increase the sample rate that is applied by the audio polyphase filters <b>606</b> and <b>614</b> to output the decoded NICAM audio data at a faster rate. Alternatively, if the FIFO <b>632</b> is less than half-full with decoded NICAM audio data, then the output rate of the decoded NICAM audio data is too fast. In this case, the NICAM sampling control block <b>634</b> can decrease the sample rate applied by the audio polyphase filters <b>606</b> and <b>614</b> to output the decoded NICAM audio data at a slower rate.

0175To initialize the operation of the NICAM processing, the length of time required for the FIFO <b>632</b> to reach “half-fullness” is measured. This time is then used to set the initial, nominal rate of resampling that is used in the audio polyphase filter blocks <b>606</b> and <b>614</b>. The measurement of the “half-fullness” time and setting of the resampling rate is performed by the NICAM sampling control block <b>634</b>. In other words, the rate at which the NICAM data is produced is measured so that the rate at which this data needs to be output can be determined. With respect to NICAM processing, the audio polyphase filter <b>628</b> has a fixed upsampling rate, and the adjustable audio polyphase filters <b>606</b> and <b>614</b> are configured to have a fixed output rate. Accordingly, changing the resampling rate of the audio polyphase filters <b>606</b> and <b>614</b> changes only the rate of their consumption of input data from the FIFO <b>632</b> in order to maintain the “half-fullness” of the FIFO <b>632</b>.

0176It should be noted that in alternative embodiments of the universal television receiver <b>100</b>, depending on the type of output signal <b>112</b> that is desired, such as just a single SIF output signal, only one input audio filtering block is needed as well as only a portion of the audio processing block <b>188</b>. Specifically, if the SIF output is all that is needed, then only the multiplexer <b>604</b> (with the SIF<b>1</b> always selected as the input), audio polyphase filter <b>606</b>, frequency rotator <b>620</b>, summer <b>624</b> (though now the input from the frequency rotator <b>622</b> is not present so the summer <b>624</b> becomes a simple pass-through block), the multiplexer <b>626</b> (with the output of the summer <b>624</b> always being the selected input) and the audio polyphase filter <b>628</b> are required. In such embodiments, the sound IF carrier signal <b>578</b> will not be provided. In this case, the audio carrier recovery can be a slave to the video carrier recovery that is performed in the video processing block <b>182</b>, as explained previously, so that once a lock is made to the video carrier signal, a similar lock can be made for the picture carrier signal. Alternatively, the first audio filtering block <b>184</b> can operate in a free running mode in which case the frequency relationship between the picture carrier signal and the audio carrier signal, which is defined by the television broadcast standard for the desired television channel signal <b>22</b>, is used to determine appropriate values for the audio frequency shift feedback signal <b>580</b> once a lock has been made for the video carrier signal.

0177The universal television receiver <b>100</b> implements a two-stage gain control technique that provides variable gain in both the analog domain (i.e. in the RF and analog processing blocks <b>102</b> and <b>104</b> (as described below)) and the digital domain (i.e. in the video processing block <b>108</b> and the audio filtering blocks <b>184</b> and <b>186</b>). The analog gain control block and the digital gain control block of the universal television receiver <b>100</b> can be considered to be the components of a gain control system that is used to control the level of analog and digital gain amplification. Conventionally, variable gain is applied only in the RF and analog processing blocks <b>102</b> and <b>104</b>. However, the universal television receiver <b>100</b> provides variable gain in both the analog and digital domains to provide another level of flexibility in gain control that results in improved signal quality in the desired television channel signal <b>22</b>.

0178The analog gain control block <b>308</b> provides at least one analog gain control signal to control an amount of analog amplification applied by at least one analog VGA in the receiver. A digital gain control block provides at least one digital gain control signal to control an amount of digital amplification applied by at least one digital VGA in the receiver. The digital gain control block can set a gain coefficient for at least one digital VGA based on a metric of the desired television channel signal <b>22</b>. The metric can be one of Signal to Noise ratio, Signal to Noise plus distortion ratio and Bit Error Rate. The metric is selected in part depending on whether the desired television channel signal <b>22</b> is transmitted according to an analog or digital broadcast standard. The digital gain control block can be block <b>362</b> or a gain control block in a digital demodulator (see <figref idref="DRAWINGS">FIG. 17</figref>). In some embodiments, the analog gain control block <b>308</b> is operable in first and second modes. In the first mode, the analog gain control block <b>308</b> generates a quasi peak measurement of a digitized version of the desired television channel signal <b>22</b> and utilizes the quasi peak measurement in a feedback loop to control the amplification of at least one analog VGA. In the second mode, the analog gain control block <b>308</b> is configured to set an initial gain coefficient of at least one analog VGA based on a metric of the desired television channel signal <b>22</b>. In alternative embodiments, the analog gain control block <b>308</b> only operates in the first mode or the second mode.

0179In the analog domain, the gain control signals provided to the various VGAs can be controlled so that the gain is distributed between the various VGAs in a more effective manner. This can be done using a variety of techniques in the first and second modes. The first mode employs a technique based on recognizing the differences in analog and digital television broadcast standards and accounting for these differences when determining an effective analog gain distribution so that the input range of the ADC <b>106</b> is effectively utilized in both cases. This technique is described in further detail with regards to <figref idref="DRAWINGS">FIGS. 16A-16C</figref>. The second mode employs a technique that uses signal and noise information, and digital and analog metrics to take distortion into account. This technique is described in further detail with regards to <figref idref="DRAWINGS">FIG. 15</figref>. In contrast, conventional techniques control two or more analog VGAs by simply using measured output levels in which the outputs contain more than just the desired television channel signal <b>22</b> and not taking into account differences in analog and digital television broadcast standards.

0180In the digital domain, digital variable gain amplification is used to ensure an adequate signal level for the desired television channel signal <b>22</b> after filtering by the video filter <b>356</b>. The amount of filtering performed by the video filter <b>356</b> is known a priori but the levels of the interferers will vary which will affect the signal level of the desired television channel signal <b>22</b>. Accordingly, the digital VGA <b>358</b> can apply gain to increase the level of the desired television channel signal <b>22</b> when operating in the analog operation mode. In embodiments in which the universal television receiver <b>100</b> is connected to a downstream digital demodulator (not shown), the demodulator can adjust the digital gain control signal <b>400</b> for appropriate amplification when the video processing block <b>182</b> is operating in digital operation mode. Although not shown, a digital VGA similar to the digital VGA <b>358</b> can be connected between the audio filter <b>564</b> and the audio polyphase filter <b>566</b> in order to compensate for the reduction in audio level due to filtering and the level of interferers that are present. In this case, audio gain control can be either independently controlled based on a desired audio level or can be slaved to the digital gain control signal <b>398</b>.

0181In an exemplary embodiment, the gain control method used herein does not rely solely on signal levels, but employs performance criteria for the demodulated video signals to more effectively set the gain settings at various locations in the RF processing block <b>102</b>, the analog processing block <b>104</b> and the digital processing block <b>108</b>. The performance criteria that are used can be the Bit-Error Rate (BER) for digital broadcast television signals, and Signal-to-Noise Ratio (SNR) or Signal-to-Noise+Distortion Ratio (SNDR) for analog broadcast television signals. Conventional gain control schemes only look at the signal strength right at the output of a variable gain amplifier and apply an amount of gain commensurate with the degree to which the signal strength is below some established level. However, it is important to note that the signal at the output of the variable gain amplifier can include more than the signal of interest and hence the measured level is not a true measure of the signal level. Accordingly, one of the gain control methods described herein controls the gain of various variable gain amplifiers by determining the signal quality of the demodulated video signal and in at least some cases can allow some level of distortion to occur assuming that the signal artifacts introduced from the distortion products do not affect the desired television channel signal <b>22</b> more than an allowable and measurable amount.

0182Referring now to <figref idref="DRAWINGS">FIG. 15</figref>, shown therein is a flow chart diagram of an exemplary embodiment of a gain control method <b>650</b> that can be employed by the universal television receiver <b>100</b> to determine the settings for the various VGAs in the RF and analog processing blocks <b>102</b> and <b>104</b>. The gain control method <b>650</b> involves performing a calibration measurement when the universal television receiver <b>100</b> is first used, and then repeating calibration thereafter from time to time to account for any changes in the environment. For example, calibration can be carried out on power-up or when changing channels. The gain control method <b>650</b> can also account for temporary interference such as planes flying close by, for example.

0183The gain control method <b>650</b> starts at step <b>652</b> in which a desired television channel is selected. The method <b>650</b> then moves to step <b>654</b> at which point a first combination of gains is applied to the VGAs in the RF and analog processing blocks <b>102</b> and <b>104</b>. At step <b>656</b>, the quality of the demodulated desired television channel signal <b>22</b> is measured using a performance metric. The performance metric can be an analog metric such as SNR or SNDR to provide information on signal amplitude and signal distortion when the desired television channel signal <b>22</b> is broadcast according to an analog standard. For example, the SNDR metric can be used to measure distortion. Alternatively, the performance metric can be a digital metric such as BER when the desired television channel signal <b>22</b> is broadcast according to a digital standard. The signal quality (SNR, SNDR, BER or the like) is measured in the digital processing block <b>108</b> and adjustments are made to the gain control signals used to control the gain of the various analog and digital variable gain amplifiers to improve this metric. When varying the gain of the VGAs at step <b>654</b>, changes to the gain coefficients are made while being careful not to overload the ADC <b>106</b>. For instance, if in step <b>654</b> the gain is increased for a VGA in an earlier stage, then a proportional reduction may be needed for a VGA at a later stage to avoid overloading the ADC <b>106</b>.

0184At step <b>658</b>, the method <b>650</b> determines whether measurements have been made for a desired set of gain value (i.e. gain settings or gain coefficients) combinations of the various VGAs. If not, the method <b>650</b> goes to step <b>654</b> to apply another combination of gain values to the VGAs in the RF and analog processing blocks <b>102</b> and <b>104</b> and the digital VGA <b>358</b>. If all of the desired gain value combinations have been tried, the method <b>650</b> then goes to step <b>660</b> in which the best combination of gain settings for the current television channel is saved in a look-up table in the memory of the digital processing block <b>108</b>. The best combination of gain settings is selected such that the input range of the ADC <b>106</b> is effectively utilized and there is an acceptable level of signal quality in the demodulated desired television channel signal <b>22</b> determined by the performance metrics and acceptable signal quality criteria (this can be obtained from the television broadcast standards). These initial gain coefficient settings for the analog VGAs can then be stored for a given television channel in a gain coefficient table. The table can be indexed according to television channel signal and during operation the gain coefficient settings can be selected from the gain coefficient look-up table based on the desired television channel signal <b>22</b>. The gain coefficient table is essentially a look-up table. At step <b>662</b>, the method <b>650</b> determines whether the gain settings for other television channels must be calibrated. If so, the method <b>650</b> goes to step <b>652</b>; if not, the method <b>650</b> goes to step <b>664</b>.

0185In an example implementation, steps <b>654</b> to <b>660</b> can involve determining a first gain coefficient for at least one VGA for a nominal desired power value for the coarse channel signal <b>162</b>, measuring a metric for the desired television channel signal <b>22</b>, repeating the setting and measuring steps for several different gain coefficients and desired power values above and below the nominal desired power value; and selecting the gain coefficient providing the best value for the metric.

0186Alternatively, instead of keeping track of the measured metrics for all gain combinations and then selecting the gain coefficients that led to the best metric, the best gain coefficients can be tracked as calibration is performed by observing if the metric decreases after a gain coefficient change; if a decrease occurs then the gain coefficient(s) can be reverted one iteration. This method of gain coefficient selection can be carried out for the gain coefficients for all VGAs or first for the gain coefficients of the VGAs in the RF stage (i.e. RF processing block) and then for the gain coefficients of the VGAs in the IF stage (i.e. analog processing block) until completed.

0187The source of the distortion due to a particular processing block is never known, thus one typically performs some iterative adjustments using this technique. Accordingly, at steps <b>654</b> to <b>658</b>, another approach can be to iterate through various gain coefficient settings for the various analog and digital VGAs and determine the maximum gain without a reduction in the measured metric. The gain coefficient for each VGA can then be adjusted upwards and downwards to determine the impact on the analog or digital metric, as the case may be, by introducing intentional distortion in order to determine how high the gain control signals can be set such that the distortion does not appreciably affect the quality of the desired television channel signal <b>22</b> after demodulation and the signal quality measured by the analog or digital metrics increase.

0188In another alternative embodiment, the current gain coefficient is obtained by measuring a first gain coefficient of a VGA to achieve a desired power value for the coarse channel signal <b>162</b>, measuring a second gain coefficient of that VGA to achieve a desired value for the metric of the desired television channel signal <b>22</b>, and calculating a difference gain coefficient or offset from the difference of the first and second gain coefficients. This VGA is now calibrated. After calibration, during use, the power at the output of the calibrated VGA can be measured, a third gain coefficient can then be calculated to achieve the desired power value and the third gain coefficient can then be adjusted by the difference gain coefficient for the calibrated VGA. The adjusted third gain coefficient is then used as the gain coefficient for the VGA. This calibration process can be performed for more than one VGA.

0189The various gain coefficient selection methods can also be performed for any digital VGAs that are used so that the gain coefficient look-up table can include gain coefficient settings for analog and digital VGAs. As before, the table is indexed according to television channel signal <b>22</b> and during operation the gain coefficient settings for the analog and digital VGAs can be selected based on the desired television channel signal <b>22</b> that is selected.

0190At step <b>664</b>, the gain settings have been calibrated for all television channels. At this point, the method <b>650</b> monitors whether there are any temporary interferers, such as an airplane that flies close by, for example. If not, the method <b>650</b> moves to step <b>668</b>. One technique for determining interference involves determining an expected power level for the desired television channel signal <b>22</b> and then monitoring the power level of the desired television channel <b>22</b> signal during use for any variations from the expected power level. If a temporary interferer is detected at step <b>664</b>, the method <b>650</b> moves to step <b>666</b> to perform an adjustment with the analog gain control settings to compensate for the temporary interference. For instance, gain adjustment may be made for a single VGA, such as the VGA <b>158</b>, to deal with a temporary interferer. The VGA <b>158</b> can be a “fine” analog VGA that has a fine step size for its gain settings while the other analog VGAs can be provided with coarse step size for their gain settings. Alternatively, more than one VGA can have fine setting control. Alternatively, one or more VGAs can be used that have a gain coefficient that is continuously variable. For a temporary interferer, the gain of the VGA <b>158</b> can be adjusted to minimize the influence of the interferer on the processed television channel signal <b>112</b>. When the temporary interference is gone, the gain setting of the VGA <b>158</b> can be set based on the look-up table and the method <b>650</b> moves to step <b>668</b>. If altering the gain coefficient for the single VGA does not provide sufficient gain variation to compensate for the detected temporary interference, then the analog gain control block adjusts the gain coefficient of at least another one of the VGAs to compensate for the detected temporary interference. The adjustment of the gain coefficients for the other VGAs can be made one at a time, that is for the first additional VGA, if adjusting the gain coefficient does not provide sufficient gain variation to compensate for the interferer then the gain coefficient for another VGA can be adjusted and so on and so forth.

0191At step <b>668</b>, the method <b>650</b> determines whether it is time to perform another calibration. If so, the method <b>650</b> goes to step <b>652</b>. If not, the method goes to step <b>664</b> and monitors for any temporary interferers. This calibration can be done in a periodic manner.

0192Accordingly, the technique of using digital and analog metrics to set the gain of the digital and analog variable gain amplifiers is used to determine the initial point at which the gain coefficients are set for the various VGAs. In this case, the initial gain coefficients are selected from the perspective of the quality of the demodulated desired television channel signal <b>22</b> rather than simply relying on power levels at the output of a VGA as is done conventionally. These gain coefficients can be used until calibration is next performed. Alternatively, the current gain coefficient for a VGA can be generated by incrementing or decrementing a previous gain coefficient based on a current measured power level of the desired television channel signal <b>22</b>. Immediately following calibration, the previous gain coefficient is the initial gain coefficient obtained from calibration.

0193As previously mentioned, the purpose of the analog gain control block <b>308</b> is to adjust the gain settings at the RF and IF stages to adjust the level of the signal presented to the ADC <b>106</b> in order to improve the quality of the desired television channel signal <b>22</b>. In an alternative, in accordance with the first mode mentioned previously, this can be done using an analog gain control feedback loop in which the output level of the ADC <b>106</b> is measured and compared to preset reference levels for both analog and digital broadcast transmission standards. If the measured level is less than the reference level then one or more gain control signals are increased, while if the measured level is higher than the reference level then one or more gain control signals are decreased. Therefore, the reference level may be thought of as a target level which the analog gain control feedback loop seeks to maintain, even when the level of the signal from the antenna <b>120</b>, or other input means as the case may be, changes or the broadcast transmission standard changes for the desired television channel signal <b>22</b>.

0194However, the characteristics of television signals transmitted under analog and digital broadcast standards, referred to herein as analog and digital broadcast television signals respectively, are fundamentally different from one another and as such the best level for digitization is different in each case. A normally (negatively) modulated analog broadcast television signal is at its highest level during the transmission of synchronizing pulses (see <figref idref="DRAWINGS">FIG. 12B</figref> for example). During these periods, the picture carrier signal resembles a pure sinusoid which has a peak to average power ratio of 3 dB. Such a signal can be optimally digitized by the ADC <b>106</b> by adjusting the amplitude of the signal such that its peaks are not clipped and a certain amount of minimum headroom is maintained with respect to the full-scale range of the ADC <b>106</b>. In practice, several dB of headroom should be provided to allow for measurement error and dynamic effects. Conversely, digital broadcast signals may have peak to average power ratios of 10-15 dB or more. When digitizing these types of signals, the amplitude of these signals should be adjusted so that excessive clipping does not occur, though some amount of clipping may be acceptable. If the output level from the ADC <b>106</b> were to be calculated based only on the average power level, then the reference level could be set to be optimal for digitizing analog broadcast television signals or for digitizing digital broadcast television signals but not for both. Furthermore, since the coarse channel signal <b>162</b> presented to the ADC <b>106</b> may simultaneously contain both analog broadcast and digital broadcast television signals, the use of average power as a measure of level is not optimal. Similarly, the use of peak power as a measurement is also not optimal.

0195Referring now to <figref idref="DRAWINGS">FIG. 16A</figref>, shown therein is a block diagram of an exemplary embodiment of the analog gain control block <b>700</b>, which employs a quasi peak detector to provide a measure of signal level for the output of the ADC <b>106</b>. The signal level measure is used to provide more effective level control when digitizing both analog and digital broadcast television signals. Furthermore, the level control is effective regardless of whether the input to the ADC <b>106</b> is dominated by the desired television channel signal <b>22</b> or an adjacent television channel signal and regardless of whether the dominant television channel signal is transmitted using analog or digital broadcast standards. The analog gain control block <b>700</b> generally tries to maintain a 7-10 dB back-off in RMS level during synchronizing intervals for analog broadcast television signals, 12 to 15 dB back-off in RMS level for digital broadcast television signals, and somewhere in between when the digitized coarse channel signal <b>172</b> from the ADC <b>106</b> includes both signal types simultaneously. The back-off is measured relative to the full scale range of the ADC <b>106</b>. Also, the analog gain control block <b>700</b> attempts to split the calculated gain between variable gain amplifiers in the RF processing block <b>102</b> and the IF variable gain amplifiers in the analog processing block <b>104</b>.

0196The analog gain control block <b>700</b> is configured to generate a quasi peak measurement to track a level substantially equal to a mean-square level based on the digitized coarse channel signal <b>172</b> during synchronizing intervals when the desired television channel signal <b>22</b> is transmitted according to an analog broadcast standard and to track a level substantially greater than the mean-square level based on the digitized coarse channel signal <b>172</b> when the desired television channel signal <b>22</b> is transmitted according to a digital broadcast standard. When the desired television channel signal <b>22</b> is transmitted according to an analog broadcast standard, the reference level is selected to provide a first amount of headroom between a root-mean-square level of the digitized coarse channel signal and the full-scale range of the ADC <b>106</b> during synchronizing intervals. When the desired television channel signal <b>22</b> is transmitted according to a digital broadcast standard, the selected reference level provides a second amount of headroom between the root-mean-square level of the digitized coarse channel signal <b>172</b> and the full-scale range of the ADC <b>106</b>. The second amount is greater than the first amount.

0197The analog gain control block <b>700</b> includes a Power Detector (PD) <b>702</b>, a lowpass filter <b>704</b> and a leaky peak detector <b>706</b> that together provide a robust measurement of the output level of the ADC <b>106</b>. The analog gain control block <b>700</b> also includes a comparator <b>708</b>, a decimation block <b>710</b>, a low pass filter <b>712</b>, a switch <b>714</b>, an IF gain adjustment path and an RF gain adjustment path. The IF gain adjustment path includes a multiplier <b>716</b>, a summer <b>718</b>, an accumulator <b>720</b> and a DAC <b>722</b>. The RF gain adjustment path includes a multiplier <b>724</b>, a summer <b>726</b>, an accumulator <b>728</b> and a DAC <b>730</b>. The analog gain control block <b>700</b> can also include an instability monitor <b>732</b> that is optional depending on the implementation of the ADC <b>106</b>. The gain adjustment path from the instability monitor <b>732</b> can also be turned off because the bandwidth of the lowpass filter <b>704</b> can be chosen such that the measurement signal from the leaky peak detector <b>706</b> is large when the ADC <b>106</b> is unstable and in this case the analog gain control block <b>700</b> is configured to reduce the analog gain control signals that it provides to bring the ADC <b>106</b> back to a stable state. However, the instability monitor <b>732</b> can be used to detect and/or reset the ADC <b>106</b> when it is unstable.

0198The switch <b>714</b> and the RF gain adjustment path can also be optional if the analog gain control block <b>700</b> is used with a receiver that has an RF processing block that is not capable of receiving a gain control signal or provides its own gain control.

0199The power detector <b>702</b> receives the digitized coarse channel signal <b>172</b> from the ADC <b>106</b> and determines the power of this signal by squaring the magnitude of the real and imaginary components of this signal. The power detector <b>702</b> provides a power signal that is filtered by the lowpass filter <b>704</b>. The lowpass filter <b>704</b> can be a first order wide-band IIR filter. The filtered power signal is then processed by the leaky peak detector <b>706</b> which tracks the peak of the output of the lowpass filter <b>704</b> and outputs a measurement signal. Since, the input signal is filtered by the lowpass filter <b>704</b> before being provided to the leaky peak detector <b>706</b>, the leaky peak detector <b>706</b> will track quasi peaks, the amplitude of which depends on the amount of lowpass filtering and the nature of the input signal, but the quasi peak is lower than the actual peak of the input signal over a period of time. Accordingly, the output of the leaky peak detector <b>706</b> can be configured to track the average power (mean-square voltage) of the digitized coarse channel signal <b>172</b> during synchronizing intervals for analog broadcast standards. The output of the leaky peak detector <b>706</b> will track a level greater than the average power of the digitized coarse channel signal <b>172</b> for digital broadcast standards.

0200Referring now to <figref idref="DRAWINGS">FIG. 16B</figref>, shown therein is a block diagram showing the functionality for an exemplary embodiment of the leaky peak detector <b>706</b>. The leaky peak detector <b>706</b> tracks the peak of its input and “leaks” over time. The leaky peak detector <b>706</b> can be implemented, from a functional point of view, using a comparator <b>733</b>, a multiplier <b>734</b>, a summer <b>736</b>, a register <b>738</b>, a switch <b>740</b>, a subtractor <b>742</b>, a register <b>744</b>, a multiplier <b>746</b> and a switch <b>748</b>. The leaky peak detector <b>706</b> also employs several parameters: a constant small decay parameter, a fast decay parameter and an attack parameter. It should be noted that the leaky peak detector <b>706</b> may be implemented using dedicated hardware or via computer code that implements the functionality of the blocks shown in <figref idref="DRAWINGS">FIG. 16B</figref> and is executed by a DSP.

0201The comparator <b>733</b> compares the level of the input signal to the leaky peak detector <b>706</b> with the previous peak value that is stored in the register <b>738</b>. If the input level is larger than the previous peak value, then the difference between the input level and the previous peak value is multiplied by an attack parameter via the multiplier <b>734</b>, added to the previous peak value by the summer <b>736</b> and provided by the switch <b>740</b> as the output of the leaky peak detector <b>706</b>. The switch <b>740</b> performs this function when notified by the comparator <b>733</b> that the input level is greater than the previous peak value. This current peak value is also stored in the register <b>738</b> as the previous peak for the next operation of the comparator <b>733</b>. The scaling of the output of the comparator <b>733</b> by the attack parameter determines how fast the leaky peak detector <b>706</b> reacts to peaks in the input signal.

0202On the other hand, when the input level is smaller than the previous peak value, the peak output value is decremented by a current decay parameter. In this case, the current decay parameter is subtracted from the previous peak value by the subtractor <b>742</b> and this reduced peak value is provided to the switch <b>740</b> which provides this reduced peak value as the output of the leaky peak detector <b>706</b>. Also, the previous peak value is updated with this reduced peak value (i.e. decremented previous peak value). The switch <b>748</b> selects either a constant small decay value or a multiplied version of a fast decay value as the current decay parameter that is stored in the register <b>744</b>. The fast decay value is multiplied by the current decay parameter stored in the register <b>744</b>. The switch <b>748</b> selects the multiplied version of the fast decay value when the input signal level is less than the previous peak value for a certain period of time, otherwise the constant small decay value is selected by the switch <b>748</b>.

0203In general, the value of the attack parameter is chosen such that the leaky peak detector <b>706</b> is reasonably responsive to peaks but not over-reactive to noise. The value of the decay parameter is chosen to be constant to avoid fluctuations due to analog broadcast video content. Since the decay parameter is small under normal operation, the leaky peak detector <b>706</b> will take a long time to decay and catch the peak of the input signal in the event of a significant drop in the amplitude of the input signal. Accordingly, a general rule for the parameters is to have fast attack and slow decay constants because the operation of the leaky peak detector <b>706</b> should be content independent (i.e. the output of the leaky peak detector <b>706</b> does not vary too much during active video lines). To improve performance, the leaky peak detector <b>706</b> is configured to enter into a fast-decay mode when it does not find a peak within a certain period of time. In the fast-decay mode, the value of the fast decay parameter is scaled recursively by the multiplier <b>746</b> until the peak output signal differs from the input signal level by a certain amount at which point the current decay value stored in the register <b>744</b> is updated with the value of the constant small decay parameter.

0204For example, for operation at 288 MHz, the decay parameter can vary in value from 0.0000004768 to 1. In at least some embodiments, the value of the decay parameter can vary from 0.0000004768 to 0.000005. In at least some embodiments, a value of 0.0000008 can be chosen for the decay parameter. The decay parameter should have a small value because the output of the leaky peak detector <b>706</b> should not fluctuate too much. The fast decay parameter can in general vary in value from 1 to 4096. In at least some embodiments, the value of the fast decay parameter can vary from 2 to 512. In at least some embodiments, for legacy (i.e. the desired television channel signal <b>22</b> is transmitted according to an analog broadcast standard) negative modulated signals a value of 256 can be chosen for fast step response. However, for legacy positive modulated signals, since the video content is above the sync tip, the fast decay parameter should have a small value, such as 2, to avoid a sudden drop in the peak value due to television signal content variation. For a desired television channel signal <b>22</b> that is transmitted according to a digital broadcast standard, a value of 256 can be chosen for the fast decay parameter for a fast step response. The attack parameter can in general vary in value from 0.000000476 to 1. In at least some embodiments, the value of the attack parameter can vary from 0.00003 to 0.02. The actual value is selected based on the input signal statistic. In at least some embodiments, for legacy signals, the leaky peak detector <b>706</b> tracks the peak so a value of 0.002 can be selected for the attack parameter. For a desired television channel signal <b>22</b> that is transmitted according to a digital broadcast standard, the leaky peak detector <b>706</b> tracks closer to the RMS value so a value of 0.0002 can be chosen for the attack parameter. It should be noted that the range of these parameters provided herein is defined by the number of bits (i.e. precision) in the digital circuit design.

0205Referring once again to <figref idref="DRAWINGS">FIG. 16A</figref>, the comparator <b>708</b> then produces a gain error signal by comparing the measurement signal with a reference level. The reference level is a target level that the analog gain control block <b>700</b> attempts to track for the output level of the ADC <b>106</b>. Since the ADC <b>106</b> operates at a high frequency, such as 288 MHz for example, the gain error signal is greatly oversampled. Furthermore, since the AGC gain control block <b>700</b> does not have to control the IF gain and RF gain at such a high rate, the gain error signal is decimated by the decimation block <b>710</b> to a much lower rate on the order of several hundred kHz. This also reduces the bit precision required for the lowpass filter <b>712</b>.

0206The reference level is selected such that during normal closed loop operation for analog broadcast television signals, the gain of the RF and IF VGAs will be adjusted to provide about 7-10 dB of back-off between the RMS level of the input signal to the ADC <b>106</b> during synchronizing intervals and the full scale range of the ADC <b>106</b>. Since the leaky peak detector <b>706</b> tracks a level greater than the average power for digital broadcast television signals, the amount of back-off between the RMS level of the input signal to the ADC <b>106</b> and the full scale range of the ADC <b>106</b> will be larger for digital broadcast television signals than analog broadcast television signals. This difference in the amount of “back-off” will be equal to the difference in tracking level between the analog and digital broadcast signals. Depending on the configuration of the low pass filter <b>704</b> and the leaky peak detector <b>706</b>, the difference may be on the order 5 dB resulting in a total back-off of 12-15 dB for digital broadcast television signals. In this way, the back-off is automatically adapted for effective digitization of the type of signal being received, without a priori knowledge of whether an analog or digital broadcast is being received. This is also effective when the coarse channel signal <b>162</b> may contain both signal types and where the relative power between them is unknown.

0207The gain error signal is then filtered by the lowpass filter <b>712</b> to produce a filtered gain error signal. However, in embodiments which include the instability monitor <b>732</b>, the analog gain control block <b>700</b> is configured to select between the gain error signal from the comparator <b>708</b> and a gain adjustment signal provided by the instability monitor <b>732</b> as the signal which is filtered by the lowpass filter <b>712</b>. In either case, the IF and RF analog gain control signals will change accordingly. In addition, the IF and RF analog gain control signals can be maintained at a previously calculated value, if this is required during operation. The operation of the instability monitor <b>732</b> is discussed in further detail below.

0208The filtered gain error signal is then scaled by an IF loop gain and accumulated in the IF gain adjustment path or scaled by an RF loop gain and accumulated in the RF gain adjustment path depending on the operation of the switch <b>714</b>. The switch <b>714</b> operates based on a take-over point and an input signal level as is shown in <figref idref="DRAWINGS">FIG. 16C</figref>. The input signal level is measured in the RF processing block <b>102</b>. At power up, i.e. initialization, both the RF and IF gain control signals are at the minimum gain level and any gain adjustments that will be made are first applied to the RF gain control signal. As the input signal level gets smaller, the level of the RF gain control signal increases until it hits a maximum level at the take-over point. Further increases in gain adjustment are then applied to the IF gain control signal. Alternatively, this figure shows that for small input signal levels, the RF gain is kept at a maximum level until the input signal level approaches the take-over point at which point the RF gain is decreased with an increase in the input signal level. The amount of IF gain is also at a maximum for weak input signals, but as the input signal level increases, the amount of IF gain is reduced until the take-over point is crossed at which point the IF gain level is held constant at a minimum level. This gain adjustment scheme allows for a maximal amount of gain to be applied earlier in the analog signal processing chain for weak input signals and a minimal amount of gain to be applied for strong input signals. The operating region to the right of the take-over point can be referred to as RF gain control mode and the region to the left of the take-over point can be referred to as IF gain control mode. Alternatively, the RF gain control can be disabled in embodiments in which the receiver uses a third party tuner that performs its own AGC regulation.

0209The combined loop gain and bandwidth of the lowpass filter <b>712</b> determines the response time of the analog gain control block <b>700</b>. The loop gain and the bandwidth of the lowpass filter <b>712</b> can be increased to improve AGC tracking for AM modulated television channel signals. However, increasing the loop gain and the bandwidth of the lowpass filter <b>712</b> too much may cause the analog gain control block <b>700</b> to be susceptible to noise or to be unstable. The loop gain for the IF gain adjustment path is provided by the multiplier <b>716</b> and the amplification factor IF loop gain. Likewise, the loop gain for the RF gain adjustment path is provided by the multiplier <b>724</b> and the amplification factor RF loop gain.

0210When the switch <b>714</b> is configured to adjust the amount of IF gain, the filtered gain error signal produced by the lowpass filter <b>712</b> is provided to the IF gain adjustment path at which point it is multiplied by the IF loop gain, and then accumulated by the summer <b>718</b> and the accumulator <b>720</b> with respect to a previous IF gain value. This can also be done in a negative fashion when the level of the analog IF gain control signal must decrease. The accumulated IF gain value is then provided to the DAC <b>722</b> to produce an analog IF gain control signal. The DAC <b>722</b> can be a 4-bit sigma-delta modulated DAC in which quantization noise is shifted to high frequencies in order to achieve high in-band bit resolution with a low-resolution DAC. Accordingly, in this case an analog low-pass filter (not shown) is also included to attenuate out-of-band noise at the output of the DAC <b>722</b>, and a 4 dB back-off is used for stability purposes.

0211When the switch <b>714</b> is configured to adjust the amount of RF gain, the filtered gain error signal produced by the lowpass filter <b>712</b> is provided to the RF gain adjustment path at which point it is multiplied by the RF loop gain, and then accumulated by the summer <b>726</b> and the accumulator <b>728</b> with respect to a previous RF gain value. This can also be done in a negative fashion when the level of the analog RF gain control signal must decrease. The accumulated RF gain value is then provided to the DAC <b>730</b> to produce an analog RF gain control signal. The DAC <b>730</b> can be a 1-bit sigma-delta modulated DAC, in which case an analog low-pass filter (not shown) is also included to attenuate out-of-band noise at the output of the DAC <b>730</b>, and a 0.45 dB back-off is used for stability purposes.

0212In embodiments of the receiver <b>100</b> that use a sigma-delta modulated ADC for the ADC <b>106</b>, the instability monitor <b>732</b> is employed to check the quantized output and analog QnOverRange and QnUnderRange status bits of the ADC <b>106</b> for detecting instability. The instability monitor <b>732</b> employs a first sliding window to check the number of times that the level of the output of the ADC <b>106</b> hits the max and min full-scale levels of the ADC <b>106</b>. The instability monitor <b>732</b> also employs a second sliding window to check the number of times that either the QnOverRange or the QnUnderRange status bit is high. These status bits indicate that the ADC <b>106</b> may be unstable. The length of each sliding window can be set based on the ratio of the sampling rate of the ADC <b>106</b> to the center frequency of the desired television channel at IF as well as to provide an indication of how many times the ADC <b>106</b> goes unstable during a given period of the desired television channel signal <b>22</b>. Each sliding window also employs a threshold with a value selected so that instability is not detected too early, i.e. due to spurious values, or too late. In some cases, a threshold value of 50% can be used.

0213When the instability monitor <b>732</b> detects instability for the ADC <b>106</b>, the analog gain processing block <b>700</b> can reduce the RF or IF gain accordingly to bring the ADC <b>106</b> to a stable state. The amount of gain adjustment is determined by the severity of the instability, which is a weighted sum of the percentage of instability indications in each sliding window described above. When the weighted instability indications exceed a programmable threshold, an instability signal will be asserted. If the RF and IF gains are reduced to the lowest levels but the instability signal is still being asserted, then the analog gain control block can output a signal to reset the ADC <b>106</b>.

0214Although <figref idref="DRAWINGS">FIG. 16A</figref> shows only one analog IF gain control signal and one analog RF gain control signal, the analog gain control block <b>700</b> can be modified to set the level of several variable gain amplifiers in the RF and analog processing blocks <b>102</b> and <b>104</b>. The analog gain control bock <b>700</b> can also be modified so that gain can be distributed amongst the digital gain amplifier used in the video processing block <b>182</b>. This can be done using more take-over points in a similar fashion as that shown in <figref idref="DRAWINGS">FIG. 16B</figref> with additional take-over points being added for switching the amount of gain control between more than one variable gain amplifier in the IF section and for switching the amount of gain control between more than one variable gain amplifier in the RF section. In other words, gain control is modified for a variable gain amplifier until it reaches a maximum/minimum setting at which point the gain control is switched to another variable gain amplifier.

0215Referring now to <figref idref="DRAWINGS">FIG. 17A</figref>, shown therein is a block diagram of another exemplary embodiment of a universal television receiver <b>750</b>. The universal television receiver <b>750</b> employs an off-the shelf RF processing block <b>752</b> which provides an IF multi-channel television signal <b>136</b>′ centered at 44 MHz (North America), 59 MHz (Japan) or 36 MHz (elsewhere). The IF multi-channel television signal <b>136</b>′ includes the desired television channel signal and at least a portion of the frequency content of one or more adjacent television channel signals. In general, the components of the receiver <b>750</b> operate in a similar fashion as the components of the universal television receiver <b>100</b> with differences explained below. For instance, the analog processing block <b>754</b> and the digital processing block <b>756</b> have a similar structure and operation compared with the corresponding blocks in the universal television receiver <b>100</b> with some changes made to operating frequency and processing methodology.

0216For the analog processing block <b>754</b>, switched capacitor filters are not used and so continuous time sub-sampling is not done. In addition, while the off-the-shelf RF processing block <b>752</b> may employ a SAW filter, coarse bandpass filters similar to coarse bandpass filters <b>150</b> and <b>156</b> are typically still needed for attenuation and anti-aliasing with the requirement that there is sufficient rolloff to provide a sufficient amount of attenuation (such as 72 dB for example) for overlapping signal components (due to sampling) near the coarse frequency region of interest. However, if the RF processing block <b>752</b> provides sufficient attenuation, then no additional filtering may be necessary. These filters in the analog processing block <b>754</b> have a center frequency at the IF frequency of the RF processing block <b>752</b>. The sampling rate used for the ADC <b>106</b> can be selected to be several times the IF frequency. For example, the sampling rate can be on the order of 288 MHz.

0217For the digital processing block <b>756</b>, depending on the sampling rate and the amount of downsampling that is used, some of the order of the components in this block, such as a frequency rotator, and the combination of a filter and a downsampler, may be reversed for improved processing efficiency. Also, in the digital processing block <b>756</b>, equalization does not have to be performed if the digital demodulator <b>758</b> provides this function. In addition, analog carrier recovery is still performed due to uncertainty in the reference frequencies used in the frequency synthesizers in the RF and analog processing blocks <b>752</b> and <b>754</b> as well as the transmitters that transmit the television signals. Furthermore, the digital processing block <b>756</b> has a video processing block corresponding to the video processing block <b>182</b>. Accordingly, if the desired television channel signal <b>22</b> is transmitted using an analog broadcast standard, the digital processing block <b>756</b> provides an output signal to the DAC block <b>110</b> which produces one or more output signals <b>112</b> depending on the particular analog broadcast standard. If the desired television channel signal <b>22</b> is transmitted using a digital broadcast standard, the digital processing block <b>756</b> provides a digital modulated video signal <b>112</b>′ which the digital demodulator <b>758</b> processes to produce a digital transport stream output <b>762</b> that can then be processed by an MPEG-2 decoder to produce video. This processing includes tracking the carrier frequency of the desired television channel signal <b>22</b>. As previously mentioned, the tracking is applied to a certain frequency such as a center carrier frequency for all of the carriers that may be used for a give digital broadcast standard.

0218The digital demodulator <b>758</b> can optionally provide a digital mode frequency shift feedback signal <b>760</b> to the first frequency rotator of the video processing block in the digital processing block <b>756</b> to adjust the frequency shift that is applied to the processed digitized coarse channel signal <b>192</b> so that it is centered about DC regardless of the frequency offset error. The digital mode frequency shift feedback signal <b>760</b> can be provided via software or hardware as is commonly known by those skilled in the art to the first frequency rotator in the video processing block. The digital demodulator <b>758</b> can update the value of the digital mode frequency shift feedback signal <b>760</b> at various times during operation. For instance, the digital demodulator <b>758</b> can update the values in the digital mode frequency shift feedback signal <b>760</b> each time the universal television receiver <b>750</b> is tuned to a different television channel. In other embodiments, the digital demodulator <b>758</b> can also update the digital mode frequency shift feedback signal <b>760</b> to account for drift in the frequency offset error due to temperature change and the like.

0219Furthermore, the two-stage gain control method can be employed by the universal television receiver <b>750</b> in which gain control is used for variable gain amplification in both the analog stage (i.e. analog circuitry) and the digital stage (i.e. digital circuitry). The analog gain control techniques discussed in relation to FIGS. <b>15</b> and <b>16</b>A-<b>16</b>C can be used. The digital demodulator <b>758</b> can also provide the digital gain control signal <b>400</b>, as described below.

0220Referring now to <figref idref="DRAWINGS">FIG. 17B</figref>, shown therein is a block diagram of an exemplary embodiment for the digital demodulator <b>758</b>. Generally, the digital demodulator <b>758</b> includes a demodulator block <b>770</b>, an error correction block <b>772</b> and a digital gain control block <b>774</b>. This general representation covers any digital demodulator. For instance, for a DVB-T digital demodulator, the demodulator block <b>770</b> is an OFDM demodulator, and the error correction block <b>772</b> includes a Viterbi decoder and a Reed-Solomon decoder. Those skilled in the art are familiar with the implementation of the demodulator block <b>770</b>, the error correction block <b>772</b> and the digital gain control block <b>774</b> for a given digital television broadcast standard.

0221The demodulator block <b>770</b> demodulates the modulated digital signal <b>112</b>′ provided by the video processing block of the digital processing block <b>756</b>. The demodulator block <b>770</b> can also lock to the carrier frequency of the desired television channel signal <b>22</b> and can optionally generate the digital mode frequency shift feedback signal <b>760</b> so that the video processing block in the digital processing block <b>756</b> can compensate for frequency offset errors when operating in digital operation mode. The output of the demodulator block <b>770</b> is then processed by the error correction block <b>772</b> to correct for any errors in the digital television channel information and produce a digital transport stream output <b>762</b>. In some cases, the error correction block <b>772</b> can also produce the signal B<b>2</b> to signify that the desired television channel signal <b>22</b> has been properly demodulated and communicate signal B<b>2</b> back to the control block <b>190</b> as explained previously with respect to <figref idref="DRAWINGS">FIG. 7</figref>.

0222The digital gain control block <b>774</b> generates the digital gain control signal <b>400</b> based on the signal quality of the input data to the digital demodulator <b>758</b>. For instance, the digital gain control block <b>774</b>, in one implementation, compares the level of the input data to the digital demodulator <b>758</b> with a desired level, and generates an appropriate value for the digital gain control signal <b>400</b> so that this input signal is either amplified or attenuated by the video processing block to achieve this level. The digital gain control block <b>774</b> is configured to determine this amount of amplification or attenuation, as the case may be, based on the input signal before it is processed by the demodulator block <b>770</b>. Alternatively, the digital gain control block <b>774</b> can generate the digital gain control signal <b>400</b> by measuring the signal quality of the modulated digital signal <b>112</b>′ using a digital metric and adjust the value of the digital gain control signal <b>400</b> to ensure that a proper signal quality is achieved. This can also include using a gain coefficient table as was described in relation to <figref idref="DRAWINGS">FIG. 15</figref>.

0223Referring now to <figref idref="DRAWINGS">FIG. 18</figref>, shown therein is another embodiment of a receiver <b>800</b>. The receiver <b>800</b> includes an analog processing block <b>802</b>, ADC <b>106</b>, a digital processing block <b>804</b>, a FIFO block <b>806</b> and DAC block <b>110</b>. The receiver <b>800</b> receives an IF signal <b>808</b> from a third-party television tuner and processes this signal to provide output signals <b>112</b> or <b>112</b>′ depending on whether the television broadcast standard is digital or analog. The input to the DAC block <b>110</b> is a digital representation of a video output for an analog television broadcast standard which is provided to the FIFO block <b>806</b> that generates the output signal <b>112</b>′ in a digital format. A similar arrangement can be used for non-modulated audio as the SIF output is at an Intermediate Frequency (IF).

0224The IF signal <b>808</b> includes the desired television channel signal <b>22</b> and other television channel signals and in conventional receivers is generally followed by a SAW filter and a fixed-gain amplifier to compensate for the loss in the SAW filter. However, in the receiver <b>800</b>, the SAW filter and fixed-gain amplifier are not required. Rather, the receiver <b>800</b> takes advantage of the differences in the standards used for intermediate-frequencies throughout the world: 44 MHz in North America, 59 MHz in Japan and 36 MHz for most of the remaining world. Accordingly, the components of the receiver <b>800</b> operate in a substantially similar manner as was described for the corresponding components in the receiver <b>750</b>, with some changes to accommodate the third party television tuner that provides the IF signal <b>808</b>. For instance, the analog processing block <b>802</b> includes an optional attenuator and a variable-gain amplifier for signal level control. In alternative implementations, the ADC <b>106</b> is implemented as a bandpass sigma-delta ADC with its input centered at one of the aforementioned IF frequencies. Otherwise the analog processing block <b>802</b> and the digital processing block <b>804</b> operate as was described for the corresponding blocks in the receiver <b>750</b>. The receiver <b>800</b> can also apply the gain control techniques of FIGS. <b>15</b> and <b>16</b>A-<b>16</b>C with modifications made, as described previously, in the event that the third party tuner does not accept an RF gain control signal. The FIFO block <b>806</b> is used to regulate the output data flow to a downstream digital element. It should be noted that the other receiver embodiments shown herein can employ a similar FIFO block for this purpose.

0225It should be noted that there can be instances in which the clock or other operational frequencies used in the receiver embodiments described herein have significant energy at a frequency region which interferes with the processing of the desired television channel signal <b>22</b> thus compromising effective SNR. This can occur during the process of converting analog signals to a digital representation. For example, with a sampling rate of 288 MHz and the desired television channel centered at 36 MHz, signals at either of 252 MHz or 324 MHz would be aliased into the desired television channel. It should be noted that the term desired television channel refers to the frequency band that includes the frequency content of the desired television channel signal <b>22</b>. Although these frequencies are far from the input frequency, the wideband nature of television systems implies that signal power could be present there.

0226There can also be coupled signals that also interfere with the desired television channel signal <b>22</b>. For example, in any combined RF/mixed-signal system, the problem of interfering signals within the chip is a constant challenge. Internal oscillators, clocks, and circuitry can generate a plethora of frequencies any one of which may create interference on its own or when combined with other frequencies within the chip. The resulting signals, often referred to as spurs, are conventionally only addressed in silicon.

0227Conventional television tuners have some degree of filtering which serves to reduce power at the high aliasing frequencies. However, this filtering has no effect on local-oscillator leakage from the tuner or the other spurious signals just described. Although the leakage power can be as large as some television channels, it is conventionally reduced by a combination of SAW filter(s) and anti-aliasing filters before the ADC <b>106</b>. However, no SAW filters are used and a minimum of anti-alias filtering is employed in the various embodiments of the analog processing block described herein; thus, the local-oscillator signal is present at full strength and may be at an aliasing frequency (i.e. a frequency that is aliased onto the frequency range that contains the desired television channel signal <b>22</b> during processing, i.e. aliased onto the desired television channel signal <b>22</b>). In this regard, even a single tone may be problematic for television signals transmitted according to certain broadcast standards.

0228To mitigate the effects of these different types of interference, it can be assumed that interference occurs at certain frequencies and if these frequencies coincide with the desired television channel, then the control block <b>190</b> can shift clock and sampling rates so that the interference is no longer in the frequency region where the desired television channel signal <b>22</b> is being processed and no longer adversely affects the processing of the desired television channel signal <b>22</b>. The adjustment in sampling rate frequency shifts an aliased version of the interferer away from the desired television channel. Resampling ratios are used to compensate for the adjusted sampling rate as described below. The shift in clock frequency affects the frequency value of the mixing signals and the sampling rate. However, because of the use of coarse filtering, coarse channel signals, and the carrier frequency recovery performed in the video and audio processing blocks, the ADC, digital processing block, DAC and the LO of the various embodiments described herein can operate together to allow the sampling rate of the entire receiver to vary. Normally, in conventional receivers, the sampling rate is fixed. However, the technique of variation in sampling rate can be employed by the receivers described herein to allow for the avoidance of potential alias signals and spurious signals generated by the various clocking domains by shifting the frequency regions that fold back or alias, due to the nature of sampling, onto the desired television channel. Other shifts can also be employed to other components of the receiver to further deal with interferers and this is discussed further below.

0229Referring now to <figref idref="DRAWINGS">FIG. 19A</figref>, shown therein is an alternative embodiment of a universal receiver <b>900</b> that employs shifts in clock frequency and sampling rate to avoid interference that can be aliased onto the desired television channel. Although this technique is described with reference to the receiver <b>900</b>, the technique can also be used with the other receiver embodiments described herein. The universal receiver <b>900</b> comprises an analog processing block <b>902</b>, an ADC <b>106</b>, a digital processing block <b>904</b>, a DAC block <b>110</b> and a FIFO <b>906</b>. These blocks generally operate in a similar fashion as the corresponding blocks in receivers <b>750</b> and <b>800</b> with the additional feature of aliasing avoidance sampling rate adjustment described below. The universal receiver <b>900</b> also includes a variable Phase Lock Loop (PLL) <b>916</b>, and the control block <b>190</b>.

0230To avoid or otherwise mitigate the effects of aliased interference, the sampling rate of the universal receiver <b>900</b> can be changed during operation. Since the sampling rate of the ADC <b>106</b> is not fixed, the digital processing block <b>904</b> is configured to operate at a sampling rate that corresponds to the modified sampling rate of the ADC <b>106</b>. This is also true for the DAC block <b>110</b>. The entire digital signal path maintains a consistent clock rate. As noted earlier, in the description of the receiver universal television <b>100</b>, near the input and output of the digital processing block <b>904</b> there are polyphase filters. In particular, the video processing block <b>182</b> includes polyphase filters <b>352</b> and <b>368</b>, the audio filtering blocks <b>184</b> and <b>186</b> include polyphase filters <b>554</b> and <b>566</b>, and the audio processing block <b>188</b> includes polyphase filters <b>606</b>, <b>614</b>, and <b>628</b>. This is also true for the digital processing block <b>904</b>. These polyphase filters <b>352</b>, <b>368</b>, <b>554</b>, <b>566</b>, <b>606</b>, <b>614</b> and <b>628</b> are used to change the effective sampling rate that is employed so that signal sampling rates internally maintain the proper ratio with respect to physical clock rates to ensure that the filtering and processing characteristics within these blocks are consistent regardless of the actual physical sampling rates that are used. Accordingly, the polyphase filters <b>352</b> and <b>554</b> at the inputs of the video processing block <b>182</b>, the audio filtering blocks <b>184</b> and <b>186</b> and the audio processing block <b>188</b> provide a first conversion in the effective sampling rate so that processing within these blocks occurs as if the physical sampling rate was never changed. The polyphase filters <b>368</b>, <b>566</b>, <b>606</b>, <b>614</b> and <b>628</b> at the outputs of these blocks <b>182</b>-<b>188</b> then apply a second conversion in the effective sampling rate to convert back to the physical sampling rate that is used by the DAC block <b>110</b> in order to properly generate the analog output signals. Alternatively, a different output rate can be used if a standard digital output data rate had to be accommodated. Accordingly, this technique of interference avoidance includes using a first input resampling ratio for the “input” polyphase filters <b>352</b> and <b>554</b> to transform the adjusted sampling rate to a nominal processing rate that was otherwise going to be used for the processing elements between the input and output polyphase filters and using at least one output resampling ratio for the “output” polyphase filters <b>368</b>, <b>566</b>, <b>606</b>, <b>614</b> and <b>628</b> to transform the nominal processing rate to the adjusted sampling rate or another sampling rate (more outputs resampling ratios can be used if the different outputs of the universal receiver <b>900</b> are to be provided at different rates).

0231Accordingly, the digital processing block <b>904</b> is configured to apply resampling ratios to compensate for the adjusted sampling rate that is used to avoid aliased interferers. This technique can be applied to aliased versions of other interferers and is not restricted to interferers generated by the local oscillator of the tuner that is used in conjunction with the receiver. For example, harmonics of the IF picture carrier of the desired television channel signal <b>22</b> as well as distortion products of various components of the receiver are predictable sources of interferers whose frequency, and the frequency of the aliased versions thereof, are known a priori and can therefore be handled with this technique.

0232In addition, it should be noted that this aliasing avoidance technique can be used with any tuner implementation and not just the tuner implementations that are described herein. For example, this technique is applicable to single conversion tuners, dual-conversion tuners and super-heterodyne tuners.

0233The technique of shifting the sampling rate to avoid aliased interference involves several aspects. Firstly, because the physical sampling rate has changed, but the desired television channel signal <b>22</b> remains at the same input frequency relative to the ADC <b>106</b>, after sampling, the desired television channel signal <b>22</b> is now further offset from its expected location in terms of normalized frequency. As has been described previously, the architecture employed for the various receiver embodiments described herein allows for some shift in the frequency location of the desired television channel. The frequency change is determined by the control block <b>190</b>. The second aspect involves altering the effective sampling rate that is employed by the polyphase filters <b>352</b> and <b>554</b> in order to frequency translate/transform signals for processing by subsequent fixed-width filtering stages when this technique is used with the frequency resampling technique discussed earlier for handling different channel bandwidths for different broadcast standards. To accomplish this, a first or input resampling ratio is applied, which is the ratio of the original sampling rate to the new desired sampling rate, to reflect the altered relative values because of the new physical sampling rate. Thirdly, at the output of the video processing block <b>182</b>, the sampling rate is returned to its original state prior to signal conversion to the analog domain by the DAC block <b>110</b> by applying a second or output resampling ratio. The second ratio is typically the inverse of the first ratio, however, another value can be used for the second resampling rate if the output rate is desired to be at another different rate.

0234Table 1 illustrates how the resampling rates are modified to provide consistent operation when the physical sample rates of the ADC and DAC are changed in order to provide various modes of interference avoidance while configured to receive an NTSC television signal, for example. All values in Table 1 are in MHz except for the ratios F_s3/F_s2and F_s5/F_s4. It can be seen that as the sample rate of the ADC <b>106</b> changes to avoid aliasing of interferers into the desired television channel, the input to the video polyphase filter <b>352</b>, after being downsampled by a factor of 8, also changes. However, the sample rate for the output of the video polyphase filter <b>352</b> is held constant because the amount of resampling provided by the video polyphase filter <b>352</b> is changed in a similar manner as the change in the sample rate of the ADC <b>106</b>. This allows the components between the video polyphase filter <b>352</b> and the video polyphase filter <b>368</b> to operate as if there had been no change to the sample rate of the ADC <b>106</b>. The video polyphase filter <b>368</b> then applies a corresponding inverted resampling ratio (the corresponding numbers in rows 4 and 6 are the inverse of one another) so that the sample rate of the output of the video polyphase filter <b>368</b> matches the sample rate of the input of the video polyphase filter <b>352</b>. Likewise, the sample rate of the input to the DAC <b>372</b> is the same as the sample rate of the output of the ADC <b>106</b>. However, as mentioned, in some cases the video polyphase filter <b>368</b> applies a different resampling ratio so that the sampling rate of the input to the DAC <b>372</b> is different from the sample rate of the output of the ADC <b>106</b>. This change in sample rate to avoid aliasing can be done independently or in addition to the change in sample rate for multi-channel processing described with respect to <figref idref="DRAWINGS">FIG. 10</figref> depending on the particular receiver architecture in which the technique of sampling rate adjustment is employed.

0235<tables id="TABLE-US-00001" num="00001"><table frame="none" colsep="0" rowsep="0" pgwide="1"><tgroup align="left" colsep="0" rowsep="0" cols="1"><colspec colname="1" colwidth="259pt" align="center" /><thead><row><entry namest="1" nameend="1" rowsep="1">TABLE 1</entry></row></thead><tbody valign="top"><row><entry namest="1" nameend="1" align="center" rowsep="1" /></row><row><entry>Exemplary Values for Resampling Ratios (all values in MHz)</entry></row></tbody></tgroup><tgroup align="left" colsep="0" rowsep="0" cols="2"><colspec colname="offset" colwidth="84pt" align="left" /><colspec colname="1" colwidth="175pt" align="center" /><tbody valign="top"><row><entry /><entry>Mode</entry></row></tbody></tgroup><tgroup align="left" colsep="0" rowsep="0" cols="6"><colspec colname="1" colwidth="84pt" align="left" /><colspec colname="2" colwidth="35pt" align="center" /><colspec colname="3" colwidth="35pt" align="center" /><colspec colname="4" colwidth="35pt" align="center" /><colspec colname="5" colwidth="35pt" align="center" /><colspec colname="6" colwidth="35pt" align="center" /><tbody valign="top"><row><entry>Sample Rate</entry><entry>1</entry><entry>2</entry><entry>3</entry><entry>4</entry><entry>5</entry></row><row><entry namest="1" nameend="6" align="center" rowsep="1" /></row></tbody></tgroup><tgroup align="left" colsep="0" rowsep="0" cols="6"><colspec colname="1" colwidth="84pt" align="left" /><colspec colname="2" colwidth="35pt" align="char" char="." /><colspec colname="3" colwidth="35pt" align="char" char="." /><colspec colname="4" colwidth="35pt" align="char" char="." /><colspec colname="5" colwidth="35pt" align="char" char="." /><colspec colname="6" colwidth="35pt" align="char" char="." /><tbody valign="top"><row><entry>Output of ADC 106 (F_s1)</entry><entry>280</entry><entry>284</entry><entry>288</entry><entry>292</entry><entry>296</entry></row><row><entry>Input to video polyphase</entry><entry>35.0</entry><entry>35.5</entry><entry>36</entry><entry>36.5</entry><entry>37.0</entry></row><row><entry>filter 352 (F_s2)</entry></row><row><entry>Output of video polyphase</entry><entry>15.16</entry><entry>15.16</entry><entry>15.16</entry><entry>15.16</entry><entry>15.16</entry></row><row><entry>filter 352 (F_s3)</entry></row><row><entry>F_s3/F_s2</entry><entry>0.4331</entry><entry>0.4270</entry><entry>0.4211</entry><entry>0.4153</entry><entry>0.4097</entry></row><row><entry>Input to video polyphase</entry><entry>15.16</entry><entry>15.16</entry><entry>15.16</entry><entry>15.16</entry><entry>15.16</entry></row><row><entry>filter 368 (F_s4)</entry></row><row><entry>Output of video polyphase</entry><entry>35.0</entry><entry>35.5</entry><entry>36.0</entry><entry>36.5</entry><entry>37.0</entry></row><row><entry>filter 368 (F_s5)</entry></row><row><entry>F_s5/F_s4</entry><entry>2.3087</entry><entry>2.3417</entry><entry>2.3747</entry><entry>2.4077</entry><entry>2.4406</entry></row><row><entry>Input to DAC 372 (F_s6)</entry><entry>280</entry><entry>284</entry><entry>288</entry><entry>292</entry><entry>296</entry></row><row><entry namest="1" nameend="6" align="center" rowsep="1" /></row></tbody></tgroup></table></tables>

0236In an exemplary embodiment, in order to shift the physical clock and sampling rates, the universal receiver <b>900</b> employs the variable PLL <b>916</b>, which can generate several fixed output frequencies based on a sampling rate control signal received from the control block <b>190</b>. To determine whether a sampling rate offset is required, the control block <b>190</b> considers the frequencies used by all of the local oscillators that are employed by the receiver <b>900</b> as well as the frequencies used by the local oscillators that are employed in circuits which are connected to the universal receiver <b>900</b>, such as an RF front end tuner. The control block <b>190</b> considers these local frequencies and harmonics of these local oscillator frequencies to be interferers. The control block <b>190</b> then considers the aliasing bands associated with a plurality of sampling rates that can be selected from. The aliasing bands are determined based on the desired television channel that is being digitized and these aliasing bands basically occur at integer multiples of the sampling rate plus or minus the IF frequency range that corresponds to the desired television channel (see <figref idref="DRAWINGS">FIGS. 19C and 19D</figref> for an example of the aliasing bands at the first integer multiple). When the control block <b>190</b> determines that at least one interferer resides in an aliasing band when using the nominal sampling rate, the control block <b>190</b> checks for other sampling rates for which the aliasing bands do not include an interferer and then calculates the required values for the resampling ratios. Accordingly, while the amplitude of the interferer is not known a priori, it is assumed that the presence of an aliased interferer within the desired television channel will have a detrimental effect on the quality of the desired television channel signal <b>22</b> and should be removed.

0237For example, the variable PLL <b>916</b> can generate a clock signal at five fixed output frequencies: 280 MHz, 284 MHz, 288 MHz, 292 MHz and 296 MHz. The nominal operating frequency can be set to the middle of the five values, i.e. 288 MHz. When the control block <b>190</b> determines that an interferer is in an aliasing band of the 288 MHz sampling rate, the control block <b>190</b> can instruct the variable PLL <b>916</b> to generate a modified clock signal for which the interferer is not in an aliasing band. The modified clock signal is used to provide the adjusted sampling rate. The modified clock signal is then provided to the ADC <b>106</b>, the digital processing block <b>904</b> and the DAC block <b>110</b>.

0238In an exemplary embodiment shown in <figref idref="DRAWINGS">FIG. 19B</figref>, the variable PLL <b>916</b> includes a Phase/Frequency Detector (PFD) <b>920</b>, a charge (Q) pump <b>922</b>, a Voltage Controlled Oscillator (VCO) <b>924</b> and a division control block <b>926</b>. The PFD <b>920</b> receives a reference clock signal and a frequency divided version of an oscillation signal that is output by the VCO <b>924</b>. The PFD <b>920</b> determines the error in phase for the frequency divided signal with respect to the reference signal, and provides this phase error to the charge pump <b>922</b>. The charge pump <b>922</b> then generates an output voltage to track the desired output frequency of the variable PLL <b>916</b>. The VCO <b>924</b> receives the output voltage of the charge pump <b>922</b> and generates the oscillation signal. The frequency of the oscillation signal is then shifted to a lower rate by the division control block <b>926</b> according to the sampling rate control signal provided from the control block <b>190</b> based on a desired sampling rate. The division control block <b>926</b> then provides the desired clock signal for sampling at the desired sampling rate.

0239Referring now to <figref idref="DRAWINGS">FIG. 19C</figref>, shown therein is a spectral plot showing the effects of aliased interference when the aliasing avoidance technique is not employed. In this example, a nominal sampling rate of 288 MHz is employed for analog-to-digital conversion and the desired television channel signal <b>22</b> is a European 8 MHz wide television channel centered at 36 MHz. The ADC <b>106</b> is also centered at 36 MHz which is ⅛^thof the sampling rate. An interfering signal is present at 323 MHz, which could be a Local-Oscillator (LO) signal from a third party tuner (i.e. RF processing block) for example, and the aliased version of this signal appears 1 MHz away from the center of the coarse region of interest at IF (as the interferer is 1 MHz away from the center of the aliasing band), which reduces the effective SNR of the desired television channel signal <b>22</b>. However, by changing the frequency of the clock signal provided by the variable PLL <b>916</b>, the sampling rate can be altered as shown in <figref idref="DRAWINGS">FIG. 19D</figref>. For instance, if the sampling rate is increased to 296 MHz, the center of the digitized coarse region of interest is now at 37 MHz, and the desired television channel remains between 32 and 40 MHz. However, the aliased version of the interfering signal at 323 MHz is now 10 MHz away from the center of the digitized coarse region of interest and thus falls at an equivalent frequency of 27 MHz. Alternatively, the sampling rate can be changed to 292 MHz in which case the digitized coarse region of interest is now centered at 36.5 MHz. The desired television channel remains between 32-40 MHz and but the aliased interferer is moved to a frequency of 31 MHz. Alternatively, the sampling rate can be changed to 280 MHz which shifts the aliased interferer to a frequency of 43 MHz.

0240In an alternative, in addition to changing the sampling rate, another modification that can be made to move an aliased interferer away from the desired television channel is to also shift the local oscillator frequency. For instance, even if the sampling rate is changed, an aliased interferer may lie on the edge of the desired television channel. In this case, since the number of sampling rates may be limited, in order to provide an additional frequency shift to move the aliased interferer away from the desired television channel, the frequency of the local oscillator can be slightly moved when the local oscillator is the cause of the interferer. The effect of shifting the frequency of the local oscillator in this way is also a shift in the IF frequency of the desired television channel. However, this shift can be tolerated due to the coarse nature of the filtering that is employed by the receiver <b>900</b> as well as the frequency locking that is employed by the video and audio processing blocks, which were described for the universal television receiver <b>100</b>.

0241In another alternative, in addition to changing the sampling rate, another modification that can be made to move an aliased interferer away from the coarse frequency region of interest is to calculate the offset of the center frequency of the desired television channel within the coarse frequency region (i.e. the capture bandwidth of the ADC <b>106</b>). This offset is available based on the frequency tracking that is done by the video processing block <b>182</b> as described with reference to <figref idref="DRAWINGS">FIG. 10</figref>. The sampling rate can then be adjusted accordingly so that the aliased interferer does not overlap with the desired television channel even though the aliased interferer may overlap with a portion of the coarse frequency region of interest.

0242In another embodiment, the receiver can be modified to address interferers that are not aliased such as those interferers due to distortion, coupling and the like with respect to the Local Oscillator (LO) used in the mixing stage. Accordingly, these types of interferers include any distortion components that can end up at the desired television channel, at the image from the LO frequency or at the IF frequency. These can be due to harmonic, intermodulation, or mixing results and can be any combination of the three. For instance, interference may be due to a distortion product in which the local oscillator of the RF processing block (i.e. tuner) is one of the frequency sources. Alternatively, the interference may result when any combination of the video carrier and/or audio carrier of the desired television channel signal <b>22</b> has generated a strong intermodulation tone or harmonic product which combine at the image frequency of the LO and are then placed onto the desired television channel signal <b>22</b>.

0243The receiver can be modified by shifting the LO frequency of the mixing stage with or without sampling rate adjustment. The LO frequency can first be shifted, without employing the technique of adjusting the sampling rate described above, to frequency shift interferers away from the desired television channel. Due to this frequency shift, the desired television channel may move, but this poses no difficulties for frequency shifts up to a certain amount due to the use of coarse filtering that is employed in the RF and analog processing blocks. Receiver architectures with traditional SAW filters cannot accommodate any such shift in LO frequency since SAW filters are very frequency specific and necessitate the precise placement of the desired television channel signal <b>22</b> to ensure that it is properly filtered without losing any information. However, the various embodiments of the receiver architectures described herein use imprecise or coarse filters that have coarse pass bands that are wider than the bandwidth of the desired television channel and can therefore accommodate a shift in frequency of the desired television channel at IF at the output of the tuner (i.e. output of the RF processing block) and at the input to the demodulator (i.e. input to the analog processing block).

0244With this technique, the local oscillator provides a variable oscillation frequency that can be shifted depending on the location of the interferer. As the LO frequency is shifted, the location of the interferer tone will be moved. If the difference in frequency between the interferer and the location of the desired television channel is small then merely adjusting the LO frequency will result in a sufficient change to move the interferer out of the desired television channel while keeping the desired television channel within the confines of the coarse pass bands of the filters used in the analog processing block. The amount of frequency shift can be known a priori based on the type of interferer. For instance, if the interferer is due to a distortion product of at least one of the picture carrier and/or audio carrier of the desired television channel signal <b>22</b> combined with the frequency location of the local oscillator or its image, then the amount of the shift in the variable oscillation frequency is that which is required to shift the interferer away from the desired television channel or its image.

0245For example, referring now to <figref idref="DRAWINGS">FIG. 20A</figref> shown therein is a spectral plot illustrating interference of a desired television channel due to distortion. In this case prior to mixing by the local oscillator, the picture and audio carriers of the desired television channel signal <b>22</b> have frequencies of 76.25 MHz and 81.75 MHz respectively. The local oscillator frequency is 115.15 MHz and after mixing the picture and audio carriers have frequencies of 38.9 and 33.4 MHz respectively (i.e. LO frequency—picture carrier frequency and LO frequency—audio carrier frequency). However, after mixing a first interferer is also present at 37.35 MHz. This interferer is due to the mixing of distortion, i.e. a harmonic of the picture carrier at 152.5 MHz (i.e. 2*76.25 MHz) has mixed with the LO frequency. There is also a second interferer present at 42.85 MHz which is due to the intermodulation of the picture and audio carriers mixing with the LO. Note that that other distortion (not shown) due to a harmonic of the audio carrier (i.e. 2*audio carrier frequency) or other combinations of the picture and audio carrier frequencies (i.e. audio carrier frequency—picture carrier frequency) may get mixed into the desired television channel by the LO depending on the frequency value of the distortion and the LO frequency. In this case the first interferer at 37.35 MHz may be problematic. It should also be noted that in this case for the ADC <b>106</b>, the sampling rate is 288 MHz, the sampling band center is at 36 MHz, the lower limit of the sampling band is 31 MHz and the upper band of the sampling limit is 41 MHz. This is shown as the ADC sampling band in <figref idref="DRAWINGS">FIG. 20A</figref>.

0246Referring now to <figref idref="DRAWINGS">FIG. 20B</figref>, shown therein is a spectral plot illustrating the avoidance of the distortion interference of <figref idref="DRAWINGS">FIG. 20A</figref> by applying a local oscillator frequency shift. In this case, the picture carrier, audio carrier and distortion are at the same frequencies of 76.25 MHz, 81.75 MHz and 152.5 MHz prior to mixing, however the LO frequency has now been shifted slightly to 113.65 MHz. Accordingly, after mixing the picture and audio carriers have frequencies of 37.4 and 31.9 MHz respectively while the first interferer now has a frequency of 38.85 MHz and the second interferer now has a frequency of 44.35 MHz. Accordingly, in this case both of the interferers have now been frequency shifted away from the desired television channel by applying a small frequency shift to the LO frequency such that no interferer directly lies within the desired television band. Also, the sampling properties of the ADC <b>106</b> have not been changed.

0247In those cases in which a much larger shift is needed in the variable oscillation frequency of the LO such that the desired television channel is no longer within the coarse pass bands of the analog processing block, the technique of adjusting the sampling rate is also used to ensure that the interferer is shifted away from the desired television channel while still capturing the desired television channel within the coarse pass bands of the coarse filters employed by the analog processing block. The sampling rate adjustment moves the center of the band of interest (i.e. that which is captured by the ADC <b>106</b>) and therefore accommodates for the resulting frequency shift in the desired television channel due to changing the oscillator frequency. Accordingly, when the LO frequency has been changed by such an amount that the desired television channel moves out of the capture bandwidth of the ADC <b>106</b>, the adjustment in sampling rate can be used to move the center of the capture bandwidth of the ADC <b>106</b> by an appropriate amount so that it now includes the desired television channel and excludes the interferer.

0248For example, referring now to <figref idref="DRAWINGS">FIG. 21A</figref> shown therein is a spectral plot illustrating interference of a desired television channel due to distortion. In this case prior to mixing by the local oscillator, the picture and audio carriers of the desired television channel signal have 22 frequencies of 69.25 MHz and 74.75 MHz respectively. The local oscillator frequency is 108.2 MHz and after mixing the picture and audio carriers have frequencies of 38.9 and 33.4 MHz respectively (i.e. LO frequency—picture carrier frequency and LO frequency—audio carrier frequency). However, after mixing interferer <b>3</b> is present at 30.35 MHz. This interferer is due to the mixing of distortion, i.e. a harmonic of the picture carrier at 138.5 MHz (i.e. 2*69.25 MHz) has mixed with the LO frequency. There is also interferer <b>4</b> present at 35.85 MHz which is due to intermodulation of the picture and audio carriers mixing with the LO. In this case interferer <b>4</b> at 35.85 MHz may be problematic. It should also be noted that in this case for the ADC <b>106</b>, the sampling rate is 288 MHz, the sampling band center is at 36 MHz, the lower limit of the sampling band is 31 MHz and the upper limit of the sampling band is 41 MHz.

0249Referring now to <figref idref="DRAWINGS">FIG. 21B</figref>, shown therein is a spectral plot illustrating the avoidance of the distortion interference of <figref idref="DRAWINGS">FIG. 21A</figref> by applying a local oscillator frequency shift. In this case, the picture carrier, audio carrier and distortion are at the same frequencies of 69.25 MHz, 74.75 MHz, 138.5 and 144 MHz prior to mixing, however the LO frequency has now been shifted slightly to 110.3 MHz. Accordingly, after mixing the picture and audio carriers have frequencies of 41 and 35.5 MHz respectively while interferer <b>3</b> now has a frequency of 28.25 MHz and interferer <b>4</b> has a frequency of 33.75 MHz. Accordingly, in this case both interferers have also been frequency shifted away from the desired television channel by applying a small frequency shift to the LO frequency such that no interferer directly lies within the desired television band. However, the television band has also shifted such that it doesn't correspond with the sampling band of the ADC <b>106</b>. Accordingly, in this case, the sampling properties of the ADC <b>106</b> are changed such that the sampling rate is now 296 MHz, the sampling band center is now at 37 MHz, the lower limit of the sampling band is now at 32 MHz and the upper limit of the sampling band is now at 42 MHz. This allows the shifted desired television channel to be properly digitized without loss of information. It should be noted that the amplitudes and frequency spacing shown in <figref idref="DRAWINGS">FIGS. 20A</figref>, <b>20</b>B, <b>21</b>A and <b>21</b>B are not shown to scale.

0250It should be noted that various aspects of processing methodology and corresponding structure have been provided herein for several different embodiments of a television receiver. Processing techniques and corresponding structure have been described for processing wideband television channel signals to obtain the video and audio information of a desired television channel signal <b>22</b> that can be transmitted according to a variety of analog or digital broadcast standards. This includes applying resampling that is configurable based on the particular broadcast standard so that a main fixed video or main fixed audio filter can be used to process television channel signals of various bandwidths. Processing techniques and corresponding structure have also been described for employing variable gain control that includes a combination of analog and digital variable gain control. Several different techniques for determining how the gain is varied during operation are provided herein. Processing techniques and corresponding structure have also been described for employing phase noise reduction to compensate for any phase noise in the desired television channel signal <b>22</b> when transmitted under an analog broadcast standard. Processing techniques and corresponding structure have also been described for using various “coarse techniques” for filtering or mixing as well as using frequency tracking to accommodate various changes due to transmission frequency or hardware as described herein. Processing techniques and corresponding structure have also been described for interference avoidance based on adjusting sampling rate, shifting certain oscillation frequencies or both adjusting sampling rate and shifting certain oscillation frequencies. Processing techniques and corresponding structure have also been described for compensating for overmodulation. Various embodiments for each of these processing techniques and corresponding structure have been described herein. It should be noted that these processing techniques and corresponding structure can all be used together in one embodiment, or various sub-combinations of these processing techniques and corresponding structure can be used as described herein when appropriate, or one or more of these techniques can be used in other receiver architectures when appropriate (i.e. “when appropriate” means that the end result is a working embodiment).

0251It should be noted that the filtering and downsampling that is performed by various blocks in the universal television receiver <b>100</b> may be realized by cascading several filters and downsamplers in series. This results in improved realization efficiency and greater processing efficiency since filters with a smaller number of filter taps can be used. Further, it should be noted that the sampling rates, the degree of downsampling and the sequence order of the various filters, downsamplers and frequency rotators can be adjusted for more efficient implementation. Also, it should be understood that the digital processing block <b>108</b> is implemented as a combination of an application specific integrated circuit along with a digital signal processor, with registers and memory and the like. Accordingly, the functionality of the blocks in the digital processing block <b>108</b> is implemented using a combination of hardware and software. It should further be understood that these various blocks in the digital processing section can be implemented with a different structure, either in hardware or software, from that shown herein as long as the same functionality is maintained. Similarly, modifications can be made to the RF and analog processing blocks as long as the basic functionality is maintained.

0252It should also be noted that the various embodiments of the receivers described herein are generally configured to process analog television broadcast standards comprising NTSC, SECAM, and PAL and digital television broadcast standards comprising ATSC, DVB-T, DMB-T and ISDB-T. Also, it should be noted that the term amplification circuitry can be interpreted to include variable gain amplifiers or amplifiers that do not have variable gain.

0253In one aspect, at least one of the embodiments described herein provides a television receiver for processing television signals to provide video and audio information for a desired television channel signal, the television signals being transmitted according to a variety of television broadcast standards. The television receiver comprises an analog processing block for filtering and amplifying a multi-channel television signal to produce a coarse channel signal; an analog to digital converter (ADC) coupled to the analog processing block for digitizing the coarse channel signal to produce a digitized coarse channel signal; and a digital processing block coupled to the ADC for processing the digitized coarse channel signal to obtain video and audio information for the desired television channel signal. For analog and digital television broadcast standards, the processing comprises re-sampling a processed version of the digitized coarse channel signal at a first new sampling rate, with respect to the sampling rate employed by the ADC, to adjust a normalized bandwidth of the desired television channel signal to generally correspond to the normalized passband of a video filter.

0254For analog television broadcast standards the digital processing block is further configured to re-sample the processed version of the digitized coarse channel signal at a second new sampling rate to adjust the normalized bandwidth of the audio information of the desired television channel signal to generally correspond to the normalized passband of an audio filter.

0255The digital processing block comprises an input filtering block for processing the digitized coarse channel signal to provide a processed digitized coarse channel signal; a video processing block configured to receive and process the processed digitized coarse channel signal to provide the video information of the desired television channel signal for analog television broadcast standards or the audio and video information of the desired television channel signal for digital television broadcast standards; and an audio processing stage configured to receive and process the processed digitized coarse channel signal or a frequency shifted version of the processed digitized coarse channel signal to provide the audio content of the desired television channel signal for analog television broadcast standards.

0256From one point of view, the video processing block comprises a frequency tracking loop configured to remove extraneous signal components, output down-shifted frequency components of the desired television channel signal and correct for frequency offset errors to lock onto a picture carrier frequency of the desired television channel signal; and a phase tracking loop configured to reduce phase noise in the desired television channel signal wherein the phase tracking loop has a high bandwidth to react quickly to phase noise.

0257From another point of view, the video processing block comprises a first frequency rotator for frequency shifting the processed digitized coarse channel signal such that frequency content of the video information of the desired television channel signal is centered at complex baseband; a video polyphase filter stage coupled to the first frequency rotator and configured to resample the output of the first frequency rotator at the first new sampling rate based on the television broadcast standard of the desired television channel signal; a video filter coupled to the video polyphase filter stage for filtering the output of the video polyphase filter; and a digital variable gain amplifier coupled to the video filter for amplifying the output of the video filter.

0258For digital television (DTV) broadcast standards, the output of the digital variable gain amplifier provides the video and audio information for the desired television channel signal.

0259In some cases the receiver further comprises a digital television demodulator for receiving and processing the output of the digital variable gain amplifier.

0260Furthermore, the digital gain amplifier can be configured to apply a gain based on a measured level of the output of the digital variable gain amplifier when the desired television channel signal is transmitted according to digital broadcast standards.

0261The digital gain amplifier can be configured to apply a gain based on a filtered version of a frequency shifted version of the output of the digital gain amplifier after correcting for phase noise when the desired television channel signal is transmitted according to analog broadcast standards.

0262The video processing block further comprises a second frequency rotator coupled to the digital variable gain amplifier for frequency shifting the output of the digital variable gain amplifier; and a picture carrier recovery block coupled to the first and second frequency rotators, the picture carrier recovery block being configured to receive the output of the second frequency rotator, provide an analog mode frequency shift feedback signal to the first frequency rotator and provide the video information of the desired television channel signal when the desired television channel signal is transmitted according to analog television broadcast standards. The analog mode frequency shift feedback signal is generated for shifting a picture carrier signal in the output of the second frequency rotator towards DC.

0263The picture carrier recovery block comprises a carrier recovery filter for filtering the output of the second frequency rotator to produce a filtered picture carrier signal; a first phase rotator coupled to the carrier recovery filter for receiving the filtered picture carrier signal and providing a first phase adjustment to produce a phase-adjusted filtered picture carrier signal; and a carrier recovery block coupled to the first phase rotator and the first frequency rotator, the carrier recovery block being configured to process the phase-adjusted filtered picture carrier signal to compensate for phase noise and produce a phase control signal that is provided to the first phase rotator to control the amount of the first phase adjustment, the carrier recovery block further being configured to generate the analog mode frequency shift feedback signal.

0264The picture carrier recovery block can further comprise a Vestigial Side Band (VSB) filter for filtering the output of the second frequency rotator to produce filtered video information; and a second phase rotator coupled to the VSB filter for receiving the filtered video information and coupled to the carrier recovery block for receiving the phase control signal to provide a second phase adjustment to the filtered video information to produce phase-adjusted video information. The video processing block can further comprise a video polyphase filter coupled to the picture carrier recovery block for receiving and resampling the phase-adjusted video information at a desired sampling rate to produce the video information of the desired television channel signal.

0265The audio processing stage comprises a first audio filtering block configured to receive and process the processed digitized coarse channel signal or the frequency-shifted version of the processed digitized coarse channel signal to provide at least one of a first intermediate audio signal and a first Sound Intermediate Frequency (SIF) signal, wherein, the audio processing stage is operational when the desired television channel signal is broadcast according to analog television broadcast standards.

0266The audio processing stage further comprises a second audio filtering block configured to receive and process the processed digitized coarse channel signal or a frequency-shifted version of the processed digitized coarse channel signal to provide a second intermediate audio signal and a second SIF signal; and an audio processing block coupled to the first and second audio filtering blocks for receiving and processing at least one of the first and second intermediate audio signals and at least one of the first and second SIF signals to produce the audio information of the desired television channel signal. Only one of the first and second audio filtering blocks is operational for analog television broadcast standards that employ one audio carrier signal and both the first and second audio filtering blocks are operational for analog television broadcast standards that employ two audio carrier signals.

0267The first audio filtering block comprises a first frequency rotator for shifting the frequency content of an input signal to the baseband; an audio polyphase filter stage coupled to the first frequency rotator for resampling the output of the first frequency rotator at a second new sampling rate; an audio filter coupled to the first audio polyphase filter, the audio filter being configured to filter the output of the audio polyphase filter stage or a modified version of the output of the audio polyphase filter stage; and an audio polyphase filter coupled to the output of the audio filter for resampling the output of the audio filter at a third new sampling rate. The input signal is the processed digitized coarse channel signal or the frequency-shifted version of the processed digitized coarse channel signal, the resampling is done at the second new sampling rate to adjust the normalized bandwidth of the audio information of the desired television channel signal to generally correspond to the normalized passband of the audio filter, the resampling is done at the third new sampling rate to produce the first SIF signal and the resampling at the first and second sampling rates are based respectively on the television broadcast standard of the desired television channel signal and a desired type of output format for the audio information.

0268The first audio filtering block further comprises a frequency demodulator coupled to the audio polyphase filter for demodulating the output of the audio polyphase filter and producing the first intermediate audio signal.

0269The first audio filtering block further comprises an audio IF carrier recovery block that is configured to receive one of the first intermediate audio signal and a sound IF carrier recovery signal, and track an audio carrier signal that corresponds to the desired television channel signal and produce an audio frequency shift feedback signal to the first frequency rotator for shifting the audio carrier frequency to DC when doing baseband demodulation and for shifting frequency content of the audio information to DC for SIF only processing.

0270The audio processing stage further comprises a NICAM processing block for receiving at least one of the first and second SIF audio signals and providing a decoded NICAM audio signal when the television broadcast standard employs the NICAM audio format; and a data structure coupled to the NICAM processing block for storing data associated with the decoded NICAM audio signal. The output rate of the decoded NICAM audio signal is adjusted based on the amount of data stored in the data structure, wherein the output rate is increased when the data structure is less than half full and the output rate is decreased when the data structure is more than half full.

0271The audio processing block further comprises a first processing pathway for receiving and processing one of the first intermediate audio signal and a decoded NICAM audio signal depending on the television broadcast standard. The first processing pathway includes a first decimation filtering block having first and second stages, the first stage providing a first amount of filtering and downsampling to a first audio sampling rate and the second stage providing a second amount of filtering and downsampling to a second audio sampling rate for the output of the first stage; and a de-emphasis filter coupled to the first decimation filtering block for applying de-emphasis to the output of the second stage. The audio processing block further comprises a pilot recovery and audio block coupled to the first stage for processing the output of the first stage; and a second processing pathway for receiving and processing one of the second intermediate audio signal and the decoded NICAM audio signal depending on the television broadcast standard. The second processing pathway includes a second decimation filtering block having parallel third and fourth stages, the third stage providing a third amount of filtering and downsampling to a third audio sampling rate and the fourth stage providing a fourth amount of filtering and downsampling to a fourth audio sampling rate for the output of the third stage; and a second de-emphasis filter coupled to the second decimation filtering block for applying de-emphasis to the output of the fourth stage.

0272For processing mono, stereo and the decoded NICAM audio signals, the audio processing block further comprises a first audio polyphase filter for receiving and resampling the output of the first de-emphasis filter; a second audio polyphase filter for receiving and resampling the output of the second de-emphasis filter; a mixture block coupled to the first and second audio polyphase filters for combining the outputs for the first and second audio polyphase filters; and a third audio polyphase filter for receiving and resampling the output of the mixture block to provide an output audio signal.

0273For processing SIF signals, the audio processing block further comprises a first audio polyphase filter for receiving and resampling the first SIF signal; a second audio polyphase filter for receiving and resampling the second SIF signal; a first frequency rotator for applying a first frequency shift to the output of the first audio polyphase filter; a second frequency rotator for applying a second frequency shift to the output of the second audio polyphase filter; a summer for combining the outputs of the first and second frequency rotators; and an audio polyphase filter for receiving and resampling the output of the summer to provide an output audio signal.

0274In a first audio processing mode, the first decimation filtering block is configured to process the first intermediate audio signal, provide Secondary Audio Program (SAP) and L-R (Left-Right) audio information via the output of the first stage, and provide an FM demodulated audio baseband signal via the output of the second stage, and wherein the pilot recovery and audio block is configured to receive the SAP and L-R audio information and provide demodulated SAP and L-R audio information to the fourth stage for filtering and downsampling to the baseband.

0275The receiver can further comprise an RF processing block coupled to the analog processing block for receiving a wideband television signal and generating the multi-channel television signal. The RF processing block comprises amplification circuitry for receiving and amplifying the wide-band television signal; a mixing stage coupled to the amplification circuitry for mixing the output of the amplification circuitry; and a variable gain amplifier coupled to the mixing stage for amplifying the output of the mixing stage to produce the multi-channel television signal.

0276In some cases, the analog processing block comprises a coarse bandpass filter for filtering the multi-channel television signal; a variable gain amplifier coupled to the first coarse bandpass filter for amplifying the output of the first coarse bandpass filter; a sample and hold circuit coupled to the first variable gain amplifier for providing a discrete-time version of the output of the first variable gain amplifier; a discrete-time coarse bandpass filter coupled to the sample and hold circuit for filtering the output of the sample and hold circuit; and a discrete-time variable gain amplifier coupled to the second coarse bandpass filter for amplifying the output of the second coarse bandpass filter to provide the coarse channel signal.

0277In some cases, the ADC is a continuous-time bandpass sigma-delta converter and the analog processing block comprises a coarse bandpass filter for filtering the multi-channel television signal; a variable gain amplifier coupled to the first coarse bandpass filter for amplifying the output of the first coarse bandpass filter; a continuous-time filter coupled to the variable gain amplifier for filtering the output of the variable gain amplifier; and a continuous-time variable gain amplifier coupled to the continuous-time filter for amplifying the output of the continuous-time filter to provide the coarse channel signal.

0278In another aspect, at least one of the embodiments described herein provides a method for processing television signals to provide video and audio information for a desired television channel signal, the television signals being transmitted according to a variety of television broadcast standards. The method comprises filtering and amplifying a multi-channel television signal to produce a coarse channel signal; digitizing the coarse channel signal to produce a digitized coarse channel signal; and processing the digitized coarse channel signal to obtain video and audio information for the desired television channel signal. For analog and digital television broadcast standards, the processing comprises re-sampling a processed version of the digitized coarse channel signal at a first new sampling rate, with respect to the sampling rate employed during digitization, to adjust a normalized bandwidth of the desired television channel signal to generally correspond to the normalized passband of a video filter.

0279For analog television broadcast standards the method further comprises re-sampling the processed version of the digitized coarse channel signal at a second new sampling rate to adjust the normalized bandwidth of the audio information of the desired television channel signal to generally correspond to the normalized passband of an audio filter.

0280Processing the digitized coarse channel signal comprises processing the digitized coarse channel signal to provide a processed digitized coarse channel signal; processing the processed digitized coarse channel signal to provide the video information of the desired television channel signal for analog television broadcast standards or the audio and video information of the desired television channel signal for digital television broadcast standards; and processing the processed digitized coarse channel signal or a frequency-shifted version of the processed digitized coarse channel signal to provide the audio content of the desired television channel signal for analog television broadcast standards.

0281From one point of view, the processing used to provide the video information comprises employing a frequency tracking loop to remove extraneous signal components, output down-shifted frequency components of the desired television channel signal and correct for frequency offset errors to lock onto a picture carrier frequency of the desired television channel signal; and employing a phase tracking loop configured to reduce phase noise in the desired television channel signal wherein the phase tracking loop has a high bandwidth to react quickly to phase noise.

0282From another point of view, the processing used to provide the video information comprises frequency-shifting the processed digitized coarse channel signal to center frequency content of the video information of the desired television channel signal at complex baseband; resampling the frequency-shifted processed digitized coarse channel signal at the first new sampling rate based on the television broadcast standard of the desired television channel signal; filtering the resampled signal with the video filter; and amplifying the output of the video filter.

0283The method further comprises obtaining the video and audio information of the desired television channel signal from the amplified signal for digital television (DTV) broadcast standards.

0284The method can further comprise providing the amplified signal to a digital television demodulator for further processing.

0285The amplification step can comprise applying a gain based on a measured level of the amplified signal when the desired television channel signal is transmitted according to digital broadcast standards.

0286Alternatively, the amplification step can comprise applying a gain after frequency shifting the amplified signal and correcting for phase noise when the desired television channel signal is transmitted according to analog broadcast standards.

0287The processing used to provide the video information further comprises frequency shifting the amplified signal; and processing the frequency-shifted amplified signal to provide an analog mode frequency shift feedback signal for the first frequency shifting step and to provide the video information of the desired television channel signal when the desired television channel signal is transmitted according to analog television broadcast standards, wherein the analog mode frequency shift feedback signal is generated for shifting a picture carrier signal in the frequency-shifted amplified signal towards DC.

0288The step of processing the frequency-shifted amplified signal comprises filtering the frequency-shifted amplified signal to produce a filtered picture carrier signal; applying a first phase adjustment to the filtered picture carrier signal to produce a phase-adjusted filtered picture carrier signal; and processing the phase-adjusted filtered picture carrier signal to compensate for phase noise and produce a phase control signal that controls the first phase adjustment.

0289The step of processing the frequency-shifted amplified signal further comprises filtering the frequency-shifted amplified signal to produce filtered video information; applying a second phase adjustment to the filtered video information to produce phase-adjusted video information, and resampling the phase-adjusted video information at a desired sampling rate to provide the video information of the desired television channel signal.

0290The method comprises processing the processed digitized coarse channel signal or the frequency-shifted version of the processed digitized coarse channel signal to provide at least one of a first intermediate audio signal and a first Sound Intermediate Frequency (SIF) signal when the desired television channel signal is broadcast according to analog television broadcast standards.

0291The method further comprises processing the processed digitized coarse channel signal or the frequency-shifted version of the processed digitized coarse channel signal to provide a second intermediate audio signal and a second SIF signal; and processing at least one of the first and second intermediate audio signals and at least one of the first and second SIF signals to produce the audio information of the desired television channel signal. The first and second intermediate audio signals and the first and second SIF signals are produced when the desired television channel signal is transmitted with an analog television broadcast standard that employs two audio carrier signals.

0292The step of processing the processed digitized coarse channel signal or the frequency-shifted version of the processed digitized to provide the first intermediate audio signal and the first SIF signal comprises frequency-shifting the frequency content of an input signal to the baseband; resampling the frequency-shifted signal at a second new sampling rate to adjust the normalized bandwidth of the audio information of the desired television channel signal to generally correspond to the normalized passband of the audio filter; filtering the resampled signal or a modified version of the resampled frequency-shifted signal with the audio filter; and resampling the filtered signal at a third new sampling rate to produce the first SIF signal. The input signal is the processed digitized coarse channel signal or the frequency-shifted version of the processed digitized coarse channel signal, and resampling at the first and second sampling rates is based respectively on the television broadcast standard of the desired television channel signal and a desired type of output format for the audio information.

0293The method further comprises demodulating the first SIF signal to produce the first intermediate audio signal.

0294The method further comprises receiving one of the first intermediate audio signal and a sound IF carrier recovery signal, and tracking an audio carrier signal that corresponds to the desired television channel signal to produce an audio frequency shift feedback signal that is used in the frequency shifting step for shifting the audio carrier frequency to DC when doing baseband demodulation and for shifting frequency content of the audio information to DC for SIF only processing.

0295The method further comprises receiving at least one of the first and second SIF audio signals and providing a decoded NICAM audio signal when the television broadcast standard employs the NICAM audio format; storing data associated with the decoded NICAM audio signal in a data structure; and controlling the output rate of the decoded NICAM audio signal based on the amount of data stored in the data structure, wherein the output rate is increased when the data structure is less than half full and the output rate is decreased when the data structure is more than half full.

0296The step of processing at least one of the first and second intermediate audio signals and at least one of the first and second SIF signals to produce the audio information of the desired television channel signal comprises processing one of the first intermediate audio signal and a decoded NICAM audio signal, depending on the television broadcast standard, by employing a first amount of filtering and downsampling to produce a first signal at a first audio sampling rate and then applying a second amount of filtering and downsampling to the first signal to produce a second signal at a second audio sampling rate; applying de-emphasis to the second signal to produce a fifth signal; processing one of the second intermediate audio signal and the decoded NICAM audio signal, depending on the television broadcast standard, by employing a third amount of filtering and downs ampling to produce a third signal at a third audio sampling rate and then applying a fourth amount of filtering and downsampling to the third signal to produce a fourth signal at a fourth audio sampling rate; and applying de-emphasis to the fourth signal to produce a sixth signal.

0297For processing mono, stereo and the decoded NICAM audio signals, the method further comprises resampling the fifth signal; resampling the sixth signal; combining the resampled fifth and sixth signals to form a combined signal; and resampling the combined signal to provide an output audio signal.

0298For processing SIF signals, the method further comprises resampling the first SIF signal; resampling the second SIF signal; applying a first frequency shift to the first resampled SIF signal; applying a second frequency shift to the second resampled SIF signal; summing the first and second frequency shifted resampled SIF signals to produce a summed signal; and resampling the summed signal to provide an output audio signal.

0299In a first audio processing mode, the step of processing one of the first intermediate audio signal and a decoded NICAM audio signal comprises processing the first intermediate audio signal to provide Secondary Audio Program (SAP) and L-R (Left-Right) audio information via the first signal, and provide an FM demodulated audio baseband signal via the second signal, demodulating the SAP and L-R audio information and applying the fourth amount of filtering and downs ampling to the demodulated SAP and L-R audio information to the baseband to produce the fourth signal at the fourth audio sampling rate.

0300In some cases, the method further comprises receiving and amplifying the wide-band television signal; mixing the amplified signal; and amplifying the mixed signal with a variable gain amplifier to produce the multi-channel television signal.

0301In some cases, the step of filtering and amplifying the multi-television channel signal comprises filtering the multi-channel television signal with a coarse bandpass filter; amplifying the filtered multi-channel television signal; employing sub-sampling on the amplified signal to provide a discrete-time signal; filtering the discrete-time signal with a discrete-time coarse bandpass filter; and amplifying the filtered discrete-time signal with a discrete-time variable gain amplifier to provide the coarse channel signal.

0302In cases in which the digitizing step comprises employing a continuous-time bandpass sigma-delta analog to digital converter, the step of filtering and amplifying the multi-television channel signal comprises filtering the multi-channel television signal with a first coarse bandpass filter; amplifying the filtered multi-channel television signal; filtering the amplified signal with a continuous-time filter to produce a second filtered signal; and amplifying the second filtered signal with a continuous-time variable gain amplifier to provide the coarse channel signal.

0303It should be understood that various modifications can be made to the embodiments described herein, without departing from these embodiments, the scope of which is defined in the appended claims.